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-rw-r--r--ChangeLog17
-rw-r--r--configure.ac1
-rw-r--r--gst/mpegaudioparse/gstmpegaudioparse.c97
-rw-r--r--gst/mpegaudioparse/gstmpegaudioparse.h12
4 files changed, 54 insertions, 73 deletions
diff --git a/ChangeLog b/ChangeLog
index 481020e6..65a0bac2 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,22 @@
2005-08-17 Wim Taymans <wim@fluendo.com>
+ * configure.ac:
+ Added mpegaudioparse
+
+ * ext/lame/gstlame.c: (gst_lame_src_getcaps),
+ (gst_lame_src_setcaps), (gst_lame_sink_setcaps),
+ (gst_lame_sink_event), (gst_lame_chain):
+ Some cleanups.
+ Fix memleak.
+
+ * gst/mpegaudioparse/gstmpegaudioparse.c:
+ (gst_mp3parse_class_init), (gst_mp3parse_init),
+ (gst_mp3parse_chain), (gst_mp3parse_change_state):
+ * gst/mpegaudioparse/gstmpegaudioparse.h:
+ Ported mpegaudioparse
+
+2005-08-17 Wim Taymans <wim@fluendo.com>
+
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open), (gst_rtspsrc_play):
Support absolute control urls too.
diff --git a/configure.ac b/configure.ac
index 45ce24f1..9ef06881 100644
--- a/configure.ac
+++ b/configure.ac
@@ -633,6 +633,7 @@ gst/fdsrc/Makefile
gst/goom/Makefile
gst/law/Makefile
gst/level/Makefile
+gst/mpegaudioparse/Makefile
gst/realmedia/Makefile
gst/rtp/Makefile
gst/rtsp/Makefile
diff --git a/gst/mpegaudioparse/gstmpegaudioparse.c b/gst/mpegaudioparse/gstmpegaudioparse.c
index de128f5d..90775ea4 100644
--- a/gst/mpegaudioparse/gstmpegaudioparse.c
+++ b/gst/mpegaudioparse/gstmpegaudioparse.c
@@ -67,8 +67,8 @@ static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass);
static void gst_mp3parse_base_init (GstMPEGAudioParseClass * klass);
static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse);
-static void gst_mp3parse_chain (GstPad * pad, GstData * _data);
-static long bpf_from_header (GstMPEGAudioParse * parse, unsigned long header);
+static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer);
+
static int head_check (unsigned long head);
static void gst_mp3parse_set_property (GObject * object, guint prop_id,
@@ -239,14 +239,18 @@ gst_mp3parse_class_init (GstMPEGAudioParseClass * klass)
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
- g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP, g_param_spec_int ("skip", "skip", "skip", G_MININT, G_MAXINT, 0, G_PARAM_READWRITE)); /* CHECKME */
- g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE, g_param_spec_int ("bitrate", "Bitrate", "Bit Rate", G_MININT, G_MAXINT, 0, G_PARAM_READABLE)); /* CHECKME */
-
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gobject_class->set_property = gst_mp3parse_set_property;
gobject_class->get_property = gst_mp3parse_get_property;
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP,
+ g_param_spec_int ("skip", "skip", "skip",
+ G_MININT, G_MAXINT, 0, G_PARAM_READWRITE));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE,
+ g_param_spec_int ("bitrate", "Bitrate", "Bit Rate",
+ G_MININT, G_MAXINT, 0, G_PARAM_READABLE));
+
gstelement_class->change_state = gst_mp3parse_change_state;
}
@@ -256,16 +260,14 @@ gst_mp3parse_init (GstMPEGAudioParse * mp3parse)
mp3parse->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&mp3_sink_template), "sink");
- gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
-
gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain);
- gst_element_set_loop_function (GST_ELEMENT (mp3parse), NULL);
+ gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
mp3parse->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&mp3_src_template), "src");
+ gst_pad_use_fixed_caps (mp3parse->srcpad);
gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad);
- gst_pad_use_explicit_caps (mp3parse->srcpad);
/*gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); */
mp3parse->partialbuf = NULL;
@@ -275,10 +277,10 @@ gst_mp3parse_init (GstMPEGAudioParse * mp3parse)
mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
}
-static void
-gst_mp3parse_chain (GstPad * pad, GstData * _data)
+/* FIXME, use adapter */
+static GstFlowReturn
+gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
{
- GstBuffer *buf = GST_BUFFER (_data);
GstMPEGAudioParse *mp3parse;
guchar *data;
glong size, offset = 0;
@@ -287,26 +289,12 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
GstBuffer *outbuf;
guint64 last_ts;
- g_return_if_fail (pad != NULL);
- g_return_if_fail (GST_IS_PAD (pad));
- g_return_if_fail (buf != NULL);
-/* g_return_if_fail(GST_IS_BUFFER(buf)); */
-
mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
GST_DEBUG ("mp3parse: received buffer of %d bytes", GST_BUFFER_SIZE (buf));
last_ts = GST_BUFFER_TIMESTAMP (buf);
- /* FIXME, do flush */
- /*
- if (mp3parse->partialbuf) {
- gst_buffer_unref(mp3parse->partialbuf);
- mp3parse->partialbuf = NULL;
- }
- mp3parse->in_flush = TRUE;
- */
-
/* if we have something left from the previous frame */
if (mp3parse->partialbuf) {
GstBuffer *newbuf;
@@ -332,15 +320,19 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
/* search for a possible start byte */
for (; ((offset < size - 4) && (data[offset] != 0xff)); offset++)
skipped++;
- if (skipped && !mp3parse->in_flush) {
+ if (skipped) {
GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes", offset, skipped);
}
/* construct the header word */
header = GST_READ_UINT32_BE (data + offset);
/* if it's a valid header, go ahead and send off the frame */
if (head_check (header)) {
- /* calculate the bpf of the frame */
- bpf = bpf_from_header (mp3parse, header);
+ guint bitrate = 0, layer = 0, rate = 0, channels = 0;
+
+ if (!(bpf = mp3_type_frame_length_from_header (header, &layer,
+ &channels, &bitrate, &rate))) {
+ g_error ("Header failed internal error");
+ }
/********************************************************************************
* robust seek support
@@ -387,18 +379,13 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
bpf);
break;
} else {
- guint bitrate, layer, rate, channels;
-
- if (!mp3_type_frame_length_from_header (header, &layer,
- &channels, &bitrate, &rate)) {
- g_error ("Header failed internal error");
- }
if (channels != mp3parse->channels ||
rate != mp3parse->rate ||
layer != mp3parse->layer || bitrate != mp3parse->bit_rate) {
GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
- gst_pad_set_explicit_caps (mp3parse->srcpad, caps);
+ gst_pad_set_caps (mp3parse->srcpad, caps);
+ gst_caps_unref (caps);
mp3parse->channels = channels;
mp3parse->layer = layer;
@@ -412,23 +399,18 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
if (mp3parse->skip == 0) {
GST_DEBUG ("mp3parse: pushing buffer of %d bytes",
GST_BUFFER_SIZE (outbuf));
- if (mp3parse->in_flush) {
- /* FIXME do some sort of flush event */
- mp3parse->in_flush = FALSE;
- }
GST_BUFFER_TIMESTAMP (outbuf) = last_ts;
+
if (mp3parse->layer == 1) {
GST_BUFFER_DURATION (outbuf) = 384 * GST_SECOND / mp3parse->rate;
} else {
GST_BUFFER_DURATION (outbuf) = 1152 * GST_SECOND / mp3parse->rate;
}
- if (GST_PAD_CAPS (mp3parse->srcpad) != NULL) {
- gst_pad_push (mp3parse->srcpad, GST_DATA (outbuf));
- } else {
- GST_DEBUG ("No capsnego yet, delaying buffer push");
- gst_buffer_unref (outbuf);
- }
+ gst_buffer_set_caps (outbuf, GST_PAD_CAPS (pad));
+
+ gst_pad_push (mp3parse->srcpad, outbuf);
+
} else {
GST_DEBUG ("mp3parse: skipping buffer of %d bytes",
GST_BUFFER_SIZE (outbuf));
@@ -438,8 +420,7 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
}
} else {
offset++;
- if (!mp3parse->in_flush)
- GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)");
+ GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)");
}
}
/* if we have processed this block and there are still */
@@ -457,19 +438,10 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
gst_buffer_unref (mp3parse->partialbuf);
mp3parse->partialbuf = NULL;
}
-}
-
-static long
-bpf_from_header (GstMPEGAudioParse * parse, unsigned long header)
-{
- guint bitrate, layer, rate, channels, length;
- if (!(length = mp3_type_frame_length_from_header (header, &layer,
- &channels, &bitrate, &rate))) {
- return 0;
- }
+ gst_object_unref (mp3parse);
- return length;
+ return GST_FLOW_OK;
}
static gboolean
@@ -561,8 +533,8 @@ static GstElementStateReturn
gst_mp3parse_change_state (GstElement * element)
{
GstMPEGAudioParse *src;
+ GstElementStateReturn result;
- g_return_val_if_fail (GST_IS_MP3PARSE (element), GST_STATE_FAILURE);
src = GST_MP3PARSE (element);
switch (GST_STATE_TRANSITION (element)) {
@@ -575,10 +547,9 @@ gst_mp3parse_change_state (GstElement * element)
break;
}
- if (GST_ELEMENT_CLASS (parent_class)->change_state)
- return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+ result = GST_ELEMENT_CLASS (parent_class)->change_state (element);
- return GST_STATE_SUCCESS;
+ return result;
}
static gboolean
diff --git a/gst/mpegaudioparse/gstmpegaudioparse.h b/gst/mpegaudioparse/gstmpegaudioparse.h
index ce8121a0..a8c2a235 100644
--- a/gst/mpegaudioparse/gstmpegaudioparse.h
+++ b/gst/mpegaudioparse/gstmpegaudioparse.h
@@ -24,11 +24,7 @@
#include <gst/gst.h>
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
+G_BEGIN_DECLS
#define GST_TYPE_MP3PARSE \
(gst_mp3parse_get_type())
@@ -62,10 +58,6 @@ struct _GstMPEGAudioParseClass {
GType gst_mp3parse_get_type(void);
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
+G_END_DECLS
#endif /* __MP3PARSE_H__ */
generated by cgit v1.2.1 (git 2.18.0) at 2024-12-28 15:33:44 +0000