From: <wt...@ke...> - 2007-03-28 18:40:36
|
CVS Root: /cvs/gstreamer Module: gst-plugins-good Changes by: wtay Date: Wed Mar 28 2007 18:40:24 UTC Log message: Based on patch by: Stefan Kost <en...@us...> * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init), (gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init), (gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property), (gst_rtp_mp4a_depay_get_property), (gst_rtp_mp4a_depay_change_state), (gst_rtp_mp4a_depay_plugin_init): * gst/rtp/gstrtpmp4adepay.h: Added MP4A-LATM depayloader. Fixes #417792. * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process): Fixup depayloader, setting codec_data, using more efficient adaptor and rtpbuffer handling. * gst/rtsp/URLS: Add url to test above. Modified files: . : ChangeLog gst/rtp : Makefile.am gstrtp.c gstrtpmp4vdepay.c gst/rtsp : URLS Added files: gst/rtp : gstrtpmp4adepay.c gstrtpmp4adepay.h Links: http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/ChangeLog.diff?r1=1.2837&r2=1.2838 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/Makefile.am.diff?r1=1.40&r2=1.41 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/gstrtp.c.diff?r1=1.36&r2=1.37 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/gstrtpmp4adepay.c?rev=1.1&content-type=text/vnd.viewcvs-markup http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/gstrtpmp4adepay.h?rev=1.1&content-type=text/vnd.viewcvs-markup http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c.diff?r1=1.20&r2=1.21 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtsp/URLS.diff?r1=1.5&r2=1.6 ====Begin Diffs==== Index: ChangeLog =================================================================== RCS file: /cvs/gstreamer/gst-plugins-good/ChangeLog,v retrieving revision 1.2837 retrieving revision 1.2838 diff -u -d -r1.2837 -r1.2838 --- ChangeLog 25 Mar 2007 15:34:41 -0000 1.2837 +++ ChangeLog 28 Mar 2007 18:40:12 -0000 1.2838 @@ -1,3 +1,27 @@ +2007-03-28 Wim Taymans <wi...@fl...> + + Based on patch by: Stefan Kost <en...@us...> + * gst/rtp/Makefile.am: + * gst/rtp/gstrtp.c: (plugin_init): + * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init), + (gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init), + (gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps), + (gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property), + (gst_rtp_mp4a_depay_get_property), + (gst_rtp_mp4a_depay_change_state), + (gst_rtp_mp4a_depay_plugin_init): + * gst/rtp/gstrtpmp4adepay.h: + Added MP4A-LATM depayloader. Fixes #417792. + * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps), + (gst_rtp_mp4v_depay_process): + Fixup depayloader, setting codec_data, using more efficient adaptor and + rtpbuffer handling. + * gst/rtsp/URLS: + Add url to test above. 2007-03-25 Wim Taymans <wi...@fl...> * gst/rtsp/gstrtspsrc.c: (find_stream_by_setup), Index: Makefile.am RCS file: /cvs/gstreamer/gst-plugins-good/gst/rtp/Makefile.am,v retrieving revision 1.40 retrieving revision 1.41 diff -u -d -r1.40 -r1.41 --- Makefile.am 18 Feb 2007 13:24:26 -0000 1.40 +++ Makefile.am 28 Mar 2007 18:40:12 -0000 1.41 @@ -21,8 +21,8 @@ gstrtph263ppay.c \ gstrtph263pay.c \ gstrtph264depay.c \ - gstrtpL16depay.c \ - gstrtpL16pay.c \ + gstrtpL16depay.c \ + gstrtpL16pay.c \ gstasteriskh263.c \ gstrtpmp2tdepay.c \ gstrtpmp2tpay.c \ @@ -30,6 +30,7 @@ gstrtpmp4vpay.c \ gstrtpmp4gdepay.c \ gstrtpmp4gpay.c \ + gstrtpmp4adepay.c \ gstrtpspeexdepay.c \ gstrtpspeexpay.c \ gstrtpsv3vdepay.c \ @@ -54,11 +55,11 @@ noinst_HEADERS = \ gstrtpL16depay.h \ gstrtpL16pay.h \ - gstrtpac3depay.h \ + gstrtpac3depay.h \ gstrtpamrdepay.h \ gstrtpamrpay.h \ gstrtpilbcdepay.h \ - gstrtpilbcpay.h \ + gstrtpilbcpay.h \ gstrtppcmadepay.h \ gstrtppcmudepay.h \ gstrtppcmupay.h \ @@ -78,6 +79,7 @@ gstrtpmp4vpay.h \ gstrtpmp4gdepay.h \ gstrtpmp4gpay.h \ + gstrtpmp4adepay.h \ gstrtpdepay.h \ gstasteriskh263.h \ gstrtpspeexdepay.h \ Index: gstrtp.c RCS file: /cvs/gstreamer/gst-plugins-good/gst/rtp/gstrtp.c,v retrieving revision 1.36 retrieving revision 1.37 diff -u -d -r1.36 -r1.37 --- gstrtp.c 18 Feb 2007 13:24:26 -0000 1.36 +++ gstrtp.c 28 Mar 2007 18:40:12 -0000 1.37 @@ -47,6 +47,7 @@ #include "gstrtpmp2tpay.h" #include "gstrtpmp4vdepay.h" #include "gstrtpmp4vpay.h" +#include "gstrtpmp4adepay.h" #include "gstrtpmp4gdepay.h" #include "gstrtpmp4gpay.h" #include "gstrtpspeexpay.h" @@ -138,6 +139,9 @@ if (!gst_rtp_mp4v_depay_plugin_init (plugin)) return FALSE; + if (!gst_rtp_mp4a_depay_plugin_init (plugin)) + return FALSE; if (!gst_rtp_mp4g_depay_plugin_init (plugin)) --- NEW FILE: gstrtpmp4adepay.c --- /* GStreamer * Copyright (C) <2007> Nokia Corporation (contact <ste...@no...>) * <2007> Wim Taymans <wi...@fl...> * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License version 2 as published by the Free Software Foundation. * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include <gst/rtp/gstrtpbuffer.h> #include <string.h> #include "gstrtpmp4adepay.h" GST_DEBUG_CATEGORY_STATIC (rtpmp4adepay_debug); #define GST_CAT_DEFAULT (rtpmp4adepay_debug) /* elementfactory information */ static const GstElementDetails gst_rtp_mp4adepay_details = GST_ELEMENT_DETAILS ("RTP packet parser", "Codec/Depayloader/Network", "Extracts MPEG4 audio from RTP packets (RFC 3016)", "Nokia Corporation (contact <ste...@no...>), " "Wim Taymans <wi...@fl...>"); /* RtpMP4ADepay signals and args */ enum { LAST_SIGNAL }; PROP_0, static GstStaticPadTemplate gst_rtp_mp4a_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/mpeg," "mpegversion = (int) 4," "framed = (boolean) false") ); static GstStaticPadTemplate gst_rtp_mp4a_depay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MP4A-LATM\"" /* All optional parameters * * "profile-level-id=[1,MAX]" * "config=" */ ) GST_BOILERPLATE (GstRtpMP4ADepay, gst_rtp_mp4a_depay, GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD); static void gst_rtp_mp4a_depay_finalize (GObject * object); static gboolean gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf); static void gst_rtp_mp4a_depay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_mp4a_depay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement * element, GstStateChange transition); static void gst_rtp_mp4a_depay_base_init (gpointer klass) GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mp4a_depay_src_template)); gst_static_pad_template_get (&gst_rtp_mp4a_depay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_mp4adepay_details); } gst_rtp_mp4a_depay_class_init (GstRtpMP4ADepayClass * klass) GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseRTPDepayloadClass *gstbasertpdepayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; gobject_class->finalize = gst_rtp_mp4a_depay_finalize; gobject_class->set_property = gst_rtp_mp4a_depay_set_property; gobject_class->get_property = gst_rtp_mp4a_depay_get_property; gstelement_class->change_state = gst_rtp_mp4a_depay_change_state; gstbasertpdepayload_class->process = gst_rtp_mp4a_depay_process; gstbasertpdepayload_class->set_caps = gst_rtp_mp4a_depay_setcaps; GST_DEBUG_CATEGORY_INIT (rtpmp4adepay_debug, "rtpmp4adepay", 0, "MPEG4 audio RTP Depayloader"); gst_rtp_mp4a_depay_init (GstRtpMP4ADepay * rtpmp4adepay, GstRtpMP4ADepayClass * klass) rtpmp4adepay->adapter = gst_adapter_new (); gst_rtp_mp4a_depay_finalize (GObject * object) GstRtpMP4ADepay *rtpmp4adepay; rtpmp4adepay = GST_RTP_MP4A_DEPAY (object); g_object_unref (rtpmp4adepay->adapter); rtpmp4adepay->adapter = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); static gboolean gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) GstStructure *structure; GstCaps *srccaps; const gchar *str; gint clock_rate = 90000; /* default */ gint object_type = 2; /* AAC LC default */ gint channels = 2; /* default */ rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload); structure = gst_caps_get_structure (caps, 0); if (gst_structure_has_field (structure, "clock-rate")) gst_structure_get_int (structure, "clock-rate", &clock_rate); depayload->clock_rate = clock_rate; if (gst_structure_has_field (structure, "object")) gst_structure_get_int (structure, "object", &object_type); srccaps = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 4, "framed", G_TYPE_BOOLEAN, FALSE, "channels", G_TYPE_INT, channels, NULL); if ((str = gst_structure_get_string (structure, "config"))) { GValue v = { 0 }; g_value_init (&v, GST_TYPE_BUFFER); if (gst_value_deserialize (&v, str)) { GstBuffer *buffer; guint8 *data; guint size; gint i; buffer = gst_value_get_buffer (&v); gst_buffer_ref (buffer); g_value_unset (&v); data = GST_BUFFER_DATA (buffer); size = GST_BUFFER_SIZE (buffer); if (size < 2) { GST_WARNING_OBJECT (depayload, "config too short (%d < 2)", size); goto bad_config; } /* Parse StreamMuxConfig according to ISO/IEC 14496-3: * * audioMuxVersion == 0 (1 bit) * allStreamsSameTimeFraming == 1 (1 bit) * numSubFrames == 0 (6 bits) * numProgram == 0 (4 bits) * numLayer == 0 (3 bits) * We only require audioMuxVersion == 0; * The remaining bit of the second byte and the rest of the bits are used * for audioSpecificConfig which we need to set in codec_info. */ if ((data[0] & 0x80) != 0x00) { GST_WARNING_OBJECT (depayload, "unknown audioMuxVersion 1"); /* shift rest of string 15 bits down */ size -= 2; for (i = 0; i < size; i++) { data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1); /* last bit, this is probably not needed. */ data[i] = ((data[i + 1] & 1) << 7); GST_BUFFER_SIZE (buffer) = size + 1; gst_caps_set_simple (srccaps, "codec_data", GST_TYPE_BUFFER, buffer, NULL); gst_buffer_unref (buffer); } else { g_warning ("cannot convert config to buffer"); } } bad_config: gst_pad_set_caps (depayload->srcpad, srccaps); gst_caps_unref (srccaps); return TRUE; static GstBuffer * gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) GstBuffer *outbuf; if (!gst_rtp_buffer_validate (buf)) goto bad_packet; /* flush remaining data on discont */ if (GST_BUFFER_IS_DISCONT (buf)) gst_adapter_clear (rtpmp4adepay->adapter); outbuf = gst_rtp_buffer_get_payload_buffer (buf); gst_adapter_push (rtpmp4adepay->adapter, outbuf); /* RTP marker bit indicates the last packet of the AudioMuxElement => create * and push a buffer */ if (gst_rtp_buffer_get_marker (buf)) { guint avail; guint latm_header_len; guint data_len; guint8 *data; avail = gst_adapter_available (rtpmp4adepay->adapter); outbuf = gst_adapter_take_buffer (rtpmp4adepay->adapter, avail); /* determine payload length and set buffer data pointer accordingly */ /* FIXME, check for overrun */ latm_header_len = 0; data_len = 0; data = GST_BUFFER_DATA (outbuf); do { data_len += data[latm_header_len]; } while (data[latm_header_len++] == 0xff); /* just a check that lengths match, possibly there can be more than one * audioMuxElement in the payload? */ if ((data_len + latm_header_len) != avail) { GST_WARNING_OBJECT (depayload, "not all payload consumed"); GST_BUFFER_SIZE (outbuf) = avail - latm_header_len; GST_BUFFER_DATA (outbuf) += latm_header_len; gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad)); GST_DEBUG ("gst_rtp_mp4a_depay_process: pushing buffer of size %d", GST_BUFFER_SIZE (outbuf)); return outbuf; return NULL; bad_packet: { GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE, ("Packet did not validate"), (NULL)); return NULL; gst_rtp_mp4a_depay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) switch (prop_id) { default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; gst_rtp_mp4a_depay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement * element, GstStateChange transition) GstStateChangeReturn ret; rtpmp4adepay = GST_RTP_MP4A_DEPAY (element); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: gst_adapter_clear (rtpmp4adepay->adapter); ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); return ret; gboolean gst_rtp_mp4a_depay_plugin_init (GstPlugin * plugin) return gst_element_register (plugin, "rtpmp4adepay", GST_RANK_NONE, GST_TYPE_RTP_MP4A_DEPAY); --- NEW FILE: gstrtpmp4adepay.h --- #ifndef __GST_RTP_MP4A_DEPAY_H__ #define __GST_RTP_MP4A_DEPAY_H__ #include <gst/gst.h> #include <gst/base/gstadapter.h> #include <gst/rtp/gstbasertpdepayload.h> G_BEGIN_DECLS #define GST_TYPE_RTP_MP4A_DEPAY \ (gst_rtp_mp4a_depay_get_type()) #define GST_RTP_MP4A_DEPAY(obj) \ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_MP4A_DEPAY,GstRtpMP4ADepay)) #define GST_RTP_MP4A_DEPAY_CLASS(klass) \ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_MP4A_DEPAY,GstRtpMP4ADepayClass)) #define GST_IS_RTP_MP4A_DEPAY(obj) \ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_MP4A_DEPAY)) #define GST_IS_RTP_MP4A_DEPAY_CLASS(klass) \ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_MP4A_DEPAY)) typedef struct _GstRtpMP4ADepay GstRtpMP4ADepay; typedef struct _GstRtpMP4ADepayClass GstRtpMP4ADepayClass; struct _GstRtpMP4ADepay GstBaseRTPDepayload depayload; GstAdapter *adapter; struct _GstRtpMP4ADepayClass GstBaseRTPDepayloadClass parent_class; gboolean gst_rtp_mp4a_depay_plugin_init (GstPlugin * plugin); G_END_DECLS #endif /* __GST_RTP_MP4A_DEPAY_H__ */ Index: gstrtpmp4vdepay.c RCS file: /cvs/gstreamer/gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c,v retrieving revision 1.20 retrieving revision 1.21 diff -u -d -r1.20 -r1.21 --- gstrtpmp4vdepay.c 24 Jan 2007 12:22:50 -0000 1.20 +++ gstrtpmp4vdepay.c 28 Mar 2007 18:40:12 -0000 1.21 @@ -39,14 +39,12 @@ /* RtpMP4VDepay signals and args */ enum { - /* FILL ME */ LAST_SIGNAL }; - ARG_0, - ARG_FREQUENCY + PROP_0, static GstStaticPadTemplate gst_rtp_mp4v_depay_src_template = @@ -152,7 +150,6 @@ static gboolean gst_rtp_mp4v_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) - GstStructure *structure; GstRtpMP4VDepay *rtpmp4vdepay; GstCaps *srccaps; @@ -163,7 +160,8 @@ structure = gst_caps_get_structure (caps, 0); - gst_structure_get_int (structure, "clock-rate", &clock_rate); + if (gst_structure_has_field (structure, "clock-rate")) + gst_structure_get_int (structure, "clock-rate", &clock_rate); depayload->clock_rate = clock_rate; srccaps = gst_caps_new_simple ("video/mpeg", @@ -181,15 +179,14 @@ gst_buffer_ref (buffer); g_value_unset (&v); - gst_buffer_set_caps (buffer, srccaps); - gst_caps_unref (srccaps); - gst_pad_push (depayload->srcpad, buffer); + gst_caps_set_simple (srccaps, + "codec_data", GST_TYPE_BUFFER, buffer, NULL); } else { g_warning ("cannot convert config to buffer"); } } + gst_pad_set_caps (depayload->srcpad, srccaps); + gst_caps_unref (srccaps); return TRUE; } @@ -205,45 +202,29 @@ if (!gst_rtp_buffer_validate (buf)) goto bad_packet; - { - gint payload_len; - guint8 *payload; - guint32 timestamp; - payload_len = gst_rtp_buffer_get_payload_len (buf); - payload = gst_rtp_buffer_get_payload (buf); + /* flush remaining data on discont */ + if (GST_BUFFER_IS_DISCONT (buf)) + gst_adapter_clear (rtpmp4vdepay->adapter); - timestamp = gst_rtp_buffer_get_timestamp (buf); + outbuf = gst_rtp_buffer_get_payload_buffer (buf); - outbuf = gst_buffer_new_and_alloc (payload_len); - memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len); + gst_adapter_push (rtpmp4vdepay->adapter, outbuf); - gst_adapter_push (rtpmp4vdepay->adapter, outbuf); + /* if this was the last packet of the VOP, create and push a buffer */ + if (gst_rtp_buffer_get_marker (buf)) { + guint avail; - /* if this was the last packet of the VOP, create and push a buffer */ - if (gst_rtp_buffer_get_marker (buf)) { - guint avail; + avail = gst_adapter_available (rtpmp4vdepay->adapter); - avail = gst_adapter_available (rtpmp4vdepay->adapter); + outbuf = gst_adapter_take_buffer (rtpmp4vdepay->adapter, avail); - outbuf = gst_buffer_new (); - GST_BUFFER_SIZE (outbuf) = avail; - GST_BUFFER_MALLOCDATA (outbuf) = - gst_adapter_take (rtpmp4vdepay->adapter, avail); - GST_BUFFER_DATA (outbuf) = GST_BUFFER_MALLOCDATA (outbuf); - gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad)); - GST_BUFFER_TIMESTAMP (outbuf) = - timestamp * GST_SECOND / depayload->clock_rate; + gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad)); - GST_DEBUG ("gst_rtp_mp4v_depay_chain: pushing buffer of size %d", - GST_BUFFER_SIZE (outbuf)); + GST_DEBUG ("gst_rtp_mp4v_depay_chain: pushing buffer of size %d", + GST_BUFFER_SIZE (outbuf)); - return outbuf; - } else { - return NULL; - } + return outbuf; return NULL; bad_packet: Index: URLS RCS file: /cvs/gstreamer/gst-plugins-good/gst/rtsp/URLS,v retrieving revision 1.5 retrieving revision 1.6 diff -u -d -r1.5 -r1.6 --- URLS 28 Feb 2007 10:06:27 -0000 1.5 +++ URLS 28 Mar 2007 18:40:12 -0000 1.6 @@ -16,6 +16,9 @@ rtsp://kmdi.utoronto.ca:555/osconf/2004_may9.1.mp4 rtsp://a2047.v1413b.c1413.g.vq.akamaistream.net/5/2047/1413/1_h264_110/1a1a1ae656c632970267e04ebd3196c428970e7ce857b81c4aab1677e445aedc3fae1b4a7bafe013/8848125_1_110.mov +MP4V-ES/MP4A-LATM + rtsp://68.251.168.13/thisislove.3gp REAL: rtsp://213.254.239.61/farm/*/encoder/tagesschau/live1high.rm rtsp://64.192.137.105:554/real.amazon-de.eu2/phononet/B/0/0/0/H/W/Y/4/K/S/01.01.rm?cloakport=80,554,7070 |
From: <wt...@ke...> - 2008-08-05 13:54:24
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CVS Root: /cvs/gstreamer Module: gst-plugins-good Changes by: wtay Date: Tue Aug 05 2008 13:54:32 UTC Log message: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_base_init), (gst_rtp_mp1s_depay_class_init), (gst_rtp_mp1s_depay_init), (gst_rtp_mp1s_depay_setcaps), (gst_rtp_mp1s_depay_process), (gst_rtp_mp1s_depay_set_property), (gst_rtp_mp1s_depay_get_property), (gst_rtp_mp1s_depay_change_state), (gst_rtp_mp1s_depay_plugin_init): * gst/rtp/gstrtpmp1sdepay.h: Add MP1S depayloader. * gst/rtsp/URLS: Some more sample rtsp streams. Modified files: . : ChangeLog gst/rtp : Makefile.am gstrtp.c gst/rtsp : URLS Added files: gst/rtp : gstrtpmp1sdepay.c gstrtpmp1sdepay.h Links: http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/ChangeLog.diff?r1=1.3641&r2=1.3642 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/Makefile.am.diff?r1=1.49&r2=1.50 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/gstrtp.c.diff?r1=1.44&r2=1.45 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/gstrtpmp1sdepay.c?rev=1.1&content-type=text/vnd.viewcvs-markup http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/gstrtpmp1sdepay.h?rev=1.1&content-type=text/vnd.viewcvs-markup http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtsp/URLS.diff?r1=1.9&r2=1.10 ====Begin Diffs==== Index: ChangeLog =================================================================== RCS file: /cvs/gstreamer/gst-plugins-good/ChangeLog,v retrieving revision 1.3641 retrieving revision 1.3642 diff -u -d -r1.3641 -r1.3642 --- ChangeLog 5 Aug 2008 08:43:43 -0000 1.3641 +++ ChangeLog 5 Aug 2008 13:54:15 -0000 1.3642 @@ -1,5 +1,22 @@ 2008-08-05 Wim Taymans <wim...@co...> + * gst/rtp/Makefile.am: + * gst/rtp/gstrtp.c: (plugin_init): + * gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_base_init), + (gst_rtp_mp1s_depay_class_init), (gst_rtp_mp1s_depay_init), + (gst_rtp_mp1s_depay_setcaps), (gst_rtp_mp1s_depay_process), + (gst_rtp_mp1s_depay_set_property), + (gst_rtp_mp1s_depay_get_property), + (gst_rtp_mp1s_depay_change_state), + (gst_rtp_mp1s_depay_plugin_init): + * gst/rtp/gstrtpmp1sdepay.h: + Add MP1S depayloader. + + * gst/rtsp/URLS: + Some more sample rtsp streams. +2008-08-05 Wim Taymans <wim...@co...> * gst/rtsp/URLS: Add another URL. Index: Makefile.am RCS file: /cvs/gstreamer/gst-plugins-good/gst/rtp/Makefile.am,v retrieving revision 1.49 retrieving revision 1.50 diff -u -d -r1.49 -r1.50 --- Makefile.am 1 Jul 2008 13:28:00 -0000 1.49 +++ Makefile.am 5 Aug 2008 13:54:17 -0000 1.50 @@ -34,6 +34,7 @@ gstrtpL16depay.c \ gstrtpL16pay.c \ gstasteriskh263.c \ + gstrtpmp1sdepay.c \ gstrtpmp2tdepay.c \ gstrtpmp2tpay.c \ gstrtpmp4vdepay.c \ @@ -95,6 +96,7 @@ gstrtph263pay.h \ gstrtph264depay.h \ gstrtph264pay.h \ + gstrtpmp1sdepay.h \ gstrtpmp2tdepay.h \ gstrtpmp2tpay.h \ gstrtpmp4vdepay.h \ Index: gstrtp.c RCS file: /cvs/gstreamer/gst-plugins-good/gst/rtp/gstrtp.c,v retrieving revision 1.44 retrieving revision 1.45 diff -u -d -r1.44 -r1.45 --- gstrtp.c 1 Jul 2008 13:28:00 -0000 1.44 +++ gstrtp.c 5 Aug 2008 13:54:18 -0000 1.45 @@ -52,6 +52,7 @@ #include "gstrtpL16depay.h" #include "gstrtpL16pay.h" #include "gstasteriskh263.h" +#include "gstrtpmp1sdepay.h" #include "gstrtpmp2tdepay.h" #include "gstrtpmp2tpay.h" #include "gstrtpmp4vdepay.h" @@ -165,6 +166,9 @@ if (!gst_asteriskh263_plugin_init (plugin)) return FALSE; + if (!gst_rtp_mp1s_depay_plugin_init (plugin)) + return FALSE; if (!gst_rtp_mp2t_depay_plugin_init (plugin)) --- NEW FILE: gstrtpmp1sdepay.c --- /* GStreamer * Copyright (C) <2008> Wim Taymans <wim...@gm...> * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include <gst/rtp/gstrtpbuffer.h> #include <string.h> #include "gstrtpmp1sdepay.h" /* elementfactory information */ static const GstElementDetails gst_rtp_mp1sdepay_details = GST_ELEMENT_DETAILS ("RTP packet depayloader", "Codec/Depayloader/Network", "Extracts MPEG1 System Streams from RTP packets (RFC 3555)", "Wim Taymans <wim...@gm...>"); /* RtpMP1SDepay signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; PROP_0, PROP_LAST static GstStaticPadTemplate gst_rtp_mp1s_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("video/mpeg,systemstream=(boolean)true") ); /* The spec says video/MP1S but I have seen streams with other/MP1S so we will * allow them both */ static GstStaticPadTemplate gst_rtp_mp1s_depay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"other\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MP1S\";" "application/x-rtp, " "media = (string) \"video\", " "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MP1S\"") GST_BOILERPLATE (GstRtpMP1SDepay, gst_rtp_mp1s_depay, GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD); static gboolean gst_rtp_mp1s_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_mp1s_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf); static void gst_rtp_mp1s_depay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_mp1s_depay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstStateChangeReturn gst_rtp_mp1s_depay_change_state (GstElement * element, GstStateChange transition); static void gst_rtp_mp1s_depay_base_init (gpointer klass) GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mp1s_depay_src_template)); gst_static_pad_template_get (&gst_rtp_mp1s_depay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_mp1sdepay_details); } gst_rtp_mp1s_depay_class_init (GstRtpMP1SDepayClass * klass) GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseRTPDepayloadClass *gstbasertpdepayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gstbasertpdepayload_class->process = gst_rtp_mp1s_depay_process; gstbasertpdepayload_class->set_caps = gst_rtp_mp1s_depay_setcaps; gobject_class->set_property = gst_rtp_mp1s_depay_set_property; gobject_class->get_property = gst_rtp_mp1s_depay_get_property; gstelement_class->change_state = gst_rtp_mp1s_depay_change_state; gst_rtp_mp1s_depay_init (GstRtpMP1SDepay * rtpmp1sdepay, GstRtpMP1SDepayClass * klass) static gboolean gst_rtp_mp1s_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) GstCaps *srccaps; GstStructure *structure; GstRtpMP1SDepay *rtpmp1sdepay; gint clock_rate = 90000; /* default */ rtpmp1sdepay = GST_RTP_MP1S_DEPAY (depayload); structure = gst_caps_get_structure (caps, 0); gst_structure_get_int (structure, "clock-rate", &clock_rate); depayload->clock_rate = clock_rate; srccaps = gst_caps_new_simple ("video/mpeg", "systemstream", G_TYPE_BOOLEAN, TRUE, NULL); gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps); gst_caps_unref (srccaps); return TRUE; static GstBuffer * gst_rtp_mp1s_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) GstBuffer *outbuf; gint payload_len; if (G_UNLIKELY (!gst_rtp_buffer_validate (buf))) goto bad_packet; payload_len = gst_rtp_buffer_get_payload_len (buf); outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 0, -1); gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad)); GST_DEBUG ("gst_rtp_mp1s_depay_chain: pushing buffer of size %d", GST_BUFFER_SIZE (outbuf)); return outbuf; /* ERRORS */ bad_packet: { GST_ELEMENT_WARNING (rtpmp1sdepay, STREAM, DECODE, (NULL), ("Packet did not validate")); return NULL; } gst_rtp_mp1s_depay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) rtpmp1sdepay = GST_RTP_MP1S_DEPAY (object); switch (prop_id) { default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; gst_rtp_mp1s_depay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) static GstStateChangeReturn gst_rtp_mp1s_depay_change_state (GstElement * element, GstStateChange transition) GstStateChangeReturn ret; rtpmp1sdepay = GST_RTP_MP1S_DEPAY (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: case GST_STATE_CHANGE_READY_TO_PAUSED: ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); case GST_STATE_CHANGE_READY_TO_NULL: return ret; gboolean gst_rtp_mp1s_depay_plugin_init (GstPlugin * plugin) return gst_element_register (plugin, "rtpmp1sdepay", GST_RANK_MARGINAL, GST_TYPE_RTP_MP1S_DEPAY); --- NEW FILE: gstrtpmp1sdepay.h --- * Copyright (C) <2006> Wim Taymans <wi...@fl...> #ifndef __GST_RTP_MP1S_DEPAY_H__ #define __GST_RTP_MP1S_DEPAY_H__ #include <gst/gst.h> #include <gst/base/gstadapter.h> #include <gst/rtp/gstbasertpdepayload.h> G_BEGIN_DECLS #define GST_TYPE_RTP_MP1S_DEPAY \ (gst_rtp_mp1s_depay_get_type()) #define GST_RTP_MP1S_DEPAY(obj) \ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_MP1S_DEPAY,GstRtpMP1SDepay)) #define GST_RTP_MP1S_DEPAY_CLASS(klass) \ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_MP1S_DEPAY,GstRtpMP1SDepayClass)) #define GST_IS_RTP_MP1S_DEPAY(obj) \ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_MP1S_DEPAY)) #define GST_IS_RTP_MP1S_DEPAY_CLASS(klass) \ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_MP1S_DEPAY)) typedef struct _GstRtpMP1SDepay GstRtpMP1SDepay; typedef struct _GstRtpMP1SDepayClass GstRtpMP1SDepayClass; struct _GstRtpMP1SDepay GstBaseRTPDepayload depayload; struct _GstRtpMP1SDepayClass GstBaseRTPDepayloadClass parent_class; gboolean gst_rtp_mp1s_depay_plugin_init (GstPlugin * plugin); G_END_DECLS #endif /* __GST_RTP_MP1S_DEPAY_H__ */ Index: URLS RCS file: /cvs/gstreamer/gst-plugins-good/gst/rtsp/URLS,v retrieving revision 1.9 retrieving revision 1.10 diff -u -d -r1.9 -r1.10 --- URLS 5 Aug 2008 08:43:45 -0000 1.9 +++ URLS 5 Aug 2008 13:54:18 -0000 1.10 @@ -33,3 +33,6 @@ AAC, interleaved: rtsp://ia300135.us.archive.org:554/0/items/uncovered_interviews/uncovered_interviews_3_256kb.mp4 +Various samples here: + http://www.jet-stream.nl/samples/mp4.html |