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configure.ac: Added mpegaudioparse
Original commit message from CVS: * 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
This commit is contained in:
Wim Taymans
parent
07dbb08aab
commit
ba5b1e6c8e
4 changed files with 54 additions and 73 deletions
17
ChangeLog
17
ChangeLog
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@ -1,3 +1,20 @@
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2005-08-17 Wim Taymans <wim@fluendo.com>
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* configure.ac:
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Added mpegaudioparse
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* ext/lame/gstlame.c: (gst_lame_src_getcaps),
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(gst_lame_src_setcaps), (gst_lame_sink_setcaps),
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(gst_lame_sink_event), (gst_lame_chain):
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Some cleanups.
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Fix memleak.
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* gst/mpegaudioparse/gstmpegaudioparse.c:
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(gst_mp3parse_class_init), (gst_mp3parse_init),
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(gst_mp3parse_chain), (gst_mp3parse_change_state):
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* gst/mpegaudioparse/gstmpegaudioparse.h:
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Ported mpegaudioparse
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2005-08-17 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open), (gst_rtspsrc_play):
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@ -633,6 +633,7 @@ gst/fdsrc/Makefile
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gst/goom/Makefile
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gst/law/Makefile
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gst/level/Makefile
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gst/mpegaudioparse/Makefile
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gst/realmedia/Makefile
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gst/rtp/Makefile
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gst/rtsp/Makefile
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@ -67,8 +67,8 @@ static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass);
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static void gst_mp3parse_base_init (GstMPEGAudioParseClass * klass);
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static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse);
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static void gst_mp3parse_chain (GstPad * pad, GstData * _data);
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static long bpf_from_header (GstMPEGAudioParse * parse, unsigned long header);
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static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer);
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static int head_check (unsigned long head);
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static void gst_mp3parse_set_property (GObject * object, guint prop_id,
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@ -239,14 +239,18 @@ gst_mp3parse_class_init (GstMPEGAudioParseClass * klass)
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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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 */
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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 */
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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gobject_class->set_property = gst_mp3parse_set_property;
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gobject_class->get_property = gst_mp3parse_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP,
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g_param_spec_int ("skip", "skip", "skip",
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G_MININT, G_MAXINT, 0, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE,
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g_param_spec_int ("bitrate", "Bitrate", "Bit Rate",
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G_MININT, G_MAXINT, 0, G_PARAM_READABLE));
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gstelement_class->change_state = gst_mp3parse_change_state;
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}
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@ -256,16 +260,14 @@ gst_mp3parse_init (GstMPEGAudioParse * mp3parse)
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mp3parse->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&mp3_sink_template), "sink");
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gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
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gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain);
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gst_element_set_loop_function (GST_ELEMENT (mp3parse), NULL);
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gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
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mp3parse->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&mp3_src_template), "src");
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gst_pad_use_fixed_caps (mp3parse->srcpad);
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gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad);
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gst_pad_use_explicit_caps (mp3parse->srcpad);
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/*gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); */
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mp3parse->partialbuf = NULL;
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@ -275,10 +277,10 @@ gst_mp3parse_init (GstMPEGAudioParse * mp3parse)
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mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
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}
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static void
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gst_mp3parse_chain (GstPad * pad, GstData * _data)
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/* FIXME, use adapter */
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static GstFlowReturn
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gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
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{
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GstBuffer *buf = GST_BUFFER (_data);
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GstMPEGAudioParse *mp3parse;
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guchar *data;
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glong size, offset = 0;
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@ -287,26 +289,12 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
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GstBuffer *outbuf;
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guint64 last_ts;
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g_return_if_fail (pad != NULL);
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g_return_if_fail (GST_IS_PAD (pad));
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g_return_if_fail (buf != NULL);
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/* g_return_if_fail(GST_IS_BUFFER(buf)); */
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mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
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GST_DEBUG ("mp3parse: received buffer of %d bytes", GST_BUFFER_SIZE (buf));
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last_ts = GST_BUFFER_TIMESTAMP (buf);
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/* FIXME, do flush */
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/*
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if (mp3parse->partialbuf) {
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gst_buffer_unref(mp3parse->partialbuf);
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mp3parse->partialbuf = NULL;
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}
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mp3parse->in_flush = TRUE;
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*/
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/* if we have something left from the previous frame */
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if (mp3parse->partialbuf) {
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GstBuffer *newbuf;
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@ -332,15 +320,19 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
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/* search for a possible start byte */
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for (; ((offset < size - 4) && (data[offset] != 0xff)); offset++)
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skipped++;
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if (skipped && !mp3parse->in_flush) {
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if (skipped) {
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GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes", offset, skipped);
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}
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/* construct the header word */
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header = GST_READ_UINT32_BE (data + offset);
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/* if it's a valid header, go ahead and send off the frame */
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if (head_check (header)) {
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/* calculate the bpf of the frame */
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bpf = bpf_from_header (mp3parse, header);
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guint bitrate = 0, layer = 0, rate = 0, channels = 0;
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if (!(bpf = mp3_type_frame_length_from_header (header, &layer,
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&channels, &bitrate, &rate))) {
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g_error ("Header failed internal error");
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}
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/********************************************************************************
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* robust seek support
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@ -387,18 +379,13 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
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bpf);
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break;
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} else {
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guint bitrate, layer, rate, channels;
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if (!mp3_type_frame_length_from_header (header, &layer,
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&channels, &bitrate, &rate)) {
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g_error ("Header failed internal error");
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}
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if (channels != mp3parse->channels ||
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rate != mp3parse->rate ||
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layer != mp3parse->layer || bitrate != mp3parse->bit_rate) {
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GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
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gst_pad_set_explicit_caps (mp3parse->srcpad, caps);
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gst_pad_set_caps (mp3parse->srcpad, caps);
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gst_caps_unref (caps);
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mp3parse->channels = channels;
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mp3parse->layer = layer;
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@ -412,23 +399,18 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
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if (mp3parse->skip == 0) {
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GST_DEBUG ("mp3parse: pushing buffer of %d bytes",
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GST_BUFFER_SIZE (outbuf));
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if (mp3parse->in_flush) {
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/* FIXME do some sort of flush event */
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mp3parse->in_flush = FALSE;
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}
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GST_BUFFER_TIMESTAMP (outbuf) = last_ts;
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if (mp3parse->layer == 1) {
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GST_BUFFER_DURATION (outbuf) = 384 * GST_SECOND / mp3parse->rate;
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} else {
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GST_BUFFER_DURATION (outbuf) = 1152 * GST_SECOND / mp3parse->rate;
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}
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if (GST_PAD_CAPS (mp3parse->srcpad) != NULL) {
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gst_pad_push (mp3parse->srcpad, GST_DATA (outbuf));
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} else {
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GST_DEBUG ("No capsnego yet, delaying buffer push");
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gst_buffer_unref (outbuf);
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}
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gst_buffer_set_caps (outbuf, GST_PAD_CAPS (pad));
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gst_pad_push (mp3parse->srcpad, outbuf);
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} else {
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GST_DEBUG ("mp3parse: skipping buffer of %d bytes",
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GST_BUFFER_SIZE (outbuf));
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@ -438,8 +420,7 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
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}
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} else {
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offset++;
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if (!mp3parse->in_flush)
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GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)");
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GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)");
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}
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}
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/* if we have processed this block and there are still */
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@ -457,19 +438,10 @@ gst_mp3parse_chain (GstPad * pad, GstData * _data)
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gst_buffer_unref (mp3parse->partialbuf);
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mp3parse->partialbuf = NULL;
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}
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}
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static long
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bpf_from_header (GstMPEGAudioParse * parse, unsigned long header)
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{
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guint bitrate, layer, rate, channels, length;
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gst_object_unref (mp3parse);
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if (!(length = mp3_type_frame_length_from_header (header, &layer,
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&channels, &bitrate, &rate))) {
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return 0;
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}
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return length;
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return GST_FLOW_OK;
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}
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static gboolean
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@ -561,8 +533,8 @@ static GstElementStateReturn
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gst_mp3parse_change_state (GstElement * element)
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{
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GstMPEGAudioParse *src;
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GstElementStateReturn result;
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g_return_val_if_fail (GST_IS_MP3PARSE (element), GST_STATE_FAILURE);
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src = GST_MP3PARSE (element);
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switch (GST_STATE_TRANSITION (element)) {
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@ -575,10 +547,9 @@ gst_mp3parse_change_state (GstElement * element)
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break;
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}
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if (GST_ELEMENT_CLASS (parent_class)->change_state)
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return GST_ELEMENT_CLASS (parent_class)->change_state (element);
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result = GST_ELEMENT_CLASS (parent_class)->change_state (element);
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return GST_STATE_SUCCESS;
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return result;
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}
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static gboolean
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@ -24,11 +24,7 @@
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#include <gst/gst.h>
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#ifdef __cplusplus
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extern "C" {
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#endif /* __cplusplus */
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G_BEGIN_DECLS
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#define GST_TYPE_MP3PARSE \
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(gst_mp3parse_get_type())
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@ -62,10 +58,6 @@ struct _GstMPEGAudioParseClass {
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GType gst_mp3parse_get_type(void);
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#ifdef __cplusplus
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}
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#endif /* __cplusplus */
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G_END_DECLS
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#endif /* __MP3PARSE_H__ */
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