From: <bu...@ke...> - 2006-04-13 03:55:27
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CVS Root: /cvs/gstreamer Module: gst-plugins-base Changes by: burger Date: Thu Apr 13 2006 03:55:24 UTC Log message: 2006-04-12 Philippe Kalaf <phi...@co...> * gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children * gst-libs/gst/rtp/gstbasertpaudiopayload.c: * gst-libs/gst/rtp/gstbasertpaudiopayload.h: New RTP audio base payloader class. Supports frame or sample based codecs Modified files: . : ChangeLog gst-libs/gst/rtp: Makefile.am gstrtpbuffer.h Added files: gst-libs/gst/rtp: gstbasertpaudiopayload.c gstbasertpaudiopayload.h Links: http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/ChangeLog.diff?r1=1.2555&r2=1.2556 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst-libs/gst/rtp/Makefile.am.diff?r1=1.5&r2=1.6 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst-libs/gst/rtp/gstbasertpaudiopayload.c?rev=1.1&content-type=text/vnd.viewcvs-markup http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst-libs/gst/rtp/gstbasertpaudiopayload.h?rev=1.1&content-type=text/vnd.viewcvs-markup http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst-libs/gst/rtp/gstrtpbuffer.h.diff?r1=1.8&r2=1.9 ====Begin Diffs==== Index: ChangeLog =================================================================== RCS file: /cvs/gstreamer/gst-plugins-base/ChangeLog,v retrieving revision 1.2555 retrieving revision 1.2556 diff -u -d -r1.2555 -r1.2556 --- ChangeLog 12 Apr 2006 11:04:53 -0000 1.2555 +++ ChangeLog 13 Apr 2006 03:55:12 -0000 1.2556 @@ -1,3 +1,11 @@ +2006-04-12 Philippe Kalaf <phi...@co...> + + * gst-libs/gst/rtp/gstrtpbuffer.h: + Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children + * gst-libs/gst/rtp/gstbasertpaudiopayload.c: + * gst-libs/gst/rtp/gstbasertpaudiopayload.h: + New RTP audio base payloader class. Supports frame or sample based codecs 2006-04-12 Thomas Vander Stichele <thomas at apestaart dot org> * configure.ac: Index: Makefile.am RCS file: /cvs/gstreamer/gst-plugins-base/gst-libs/gst/rtp/Makefile.am,v retrieving revision 1.5 retrieving revision 1.6 diff -u -d -r1.5 -r1.6 --- Makefile.am 27 Nov 2005 16:27:20 -0000 1.5 +++ Makefile.am 13 Apr 2006 03:55:12 -0000 1.6 @@ -2,12 +2,14 @@ libgstrtpinclude_HEADERS = gstrtpbuffer.h \ gstbasertppayload.h \ + gstbasertpaudiopayload.h \ gstbasertpdepayload.h lib_LTLIBRARIES = libgstrtp-@GST_MAJORMINOR@.la libgstrtp_@GST_MAJORMINOR@_la_SOURCES = gstrtpbuffer.c \ gstbasertppayload.c \ + gstbasertpaudiopayload.c \ gstbasertpdepayload.c libgstrtp_@GST_MAJORMINOR@_la_CFLAGS = $(GST_CFLAGS) --- NEW FILE: gstbasertpaudiopayload.c --- /* GStreamer * Copyright (C) <2006> Philippe Khalaf <bu...@sp...> * * 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 <stdlib.h> #include <string.h> #include <gst/rtp/gstrtpbuffer.h> #include <math.h> #include "gstbasertpaudiopayload.h" GST_DEBUG_CATEGORY (basertpaudiopayload_debug); #define GST_CAT_DEFAULT (basertpaudiopayload_debug) /* let us define a minimum of 10 ms for sample based codecs */ #define GST_RTP_MIN_PTIME_MS 10 static void gst_basertpaudiopayload_finalize (GObject * object); static GstFlowReturn gst_basertpaudiopayload_push (GstBaseRTPPayload * basepayload, guint8 * data, guint payload_len, GstClockTime timestamp); static GstFlowReturn gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer); gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload * GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_basertpaudiopayload, GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD); static void gst_basertpaudiopayload_base_init (gpointer klass) { } gst_basertpaudiopayload_class_init (GstBaseRTPAudioPayloadClass * klass) GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; gobject_class->finalize = gst_basertpaudiopayload_finalize; parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD); gstbasertppayload_class->handle_buffer = gst_basertpaudiopayload_handle_buffer; GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0, "base audio RTP payloader"); gst_basertpaudiopayload_init (GstBaseRTPAudioPayload * basertpaudiopayload, GstBaseRTPAudioPayloadClass * klass) basertpaudiopayload->adapter = gst_adapter_new (); basertpaudiopayload->adapter_base_ts = 0; basertpaudiopayload->type = AUDIO_CODEC_TYPE_NONE; /* these need to be set by child object if frame based */ basertpaudiopayload->frame_size = 0; basertpaudiopayload->frame_duration = 0; /* these need to be set by child object if sample based */ basertpaudiopayload->sample_size = 0; gst_basertpaudiopayload_finalize (GObject * object) GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object); g_object_unref (basertpaudiopayload->adapter); basertpaudiopayload->adapter = NULL; GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object)); void gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload * basertpaudiopayload) g_return_if_fail (basertpaudiopayload != NULL); if (basertpaudiopayload->type != AUDIO_CODEC_TYPE_NONE) { GST_ERROR_OBJECT (basertpaudiopayload, "Codec type already set! You should only set this once!"); } basertpaudiopayload->type = AUDIO_CODEC_TYPE_FRAME_BASED; gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload * basertpaudiopayload->type = AUDIO_CODEC_TYPE_SAMPLE_BASED; /* These are options that need to be set for frame based audio codecs */ gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload * basertpaudiopayload, gint frame_duration, gint frame_size) basertpaudiopayload->frame_size = frame_size; basertpaudiopayload->frame_duration = frame_duration; gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload * basertpaudiopayload, gint sample_size) basertpaudiopayload->sample_size = sample_size; gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) GstFlowReturn ret; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); ret = GST_FLOW_ERROR; if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_FRAME_BASED) { ret = gst_basertpaudiopayload_handle_frame_based_buffer (basepayload, buffer); } else if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) { ret = gst_basertpaudiopayload_handle_sample_based_buffer (basepayload, } else { GST_DEBUG_OBJECT (basertpaudiopayload, "Audio codec type not set"); return ret; /* this assumes all frames have a constant duration and a constant size */ basepayload, GstBuffer * buffer) guint payload_len; guint8 *data; guint available; gint frame_size, frame_duration; guint maxptime_octets = G_MAXUINT; if (basertpaudiopayload->frame_size == 0 || basertpaudiopayload->frame_duration == 0) { GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set"); gst_buffer_unref (buffer); return GST_FLOW_ERROR; frame_size = basertpaudiopayload->frame_size; frame_duration = basertpaudiopayload->frame_duration; /* If buffer fits on an RTP packet, let's just push it through without using * the adapter */ /* this will check again max_ptime and max_mtu */ if (!gst_basertppayload_is_filled (basepayload, gst_rtp_buffer_calc_packet_len (GST_BUFFER_SIZE (buffer), 0, 0), GST_BUFFER_DURATION (buffer))) { ret = gst_basertpaudiopayload_push (basepayload, GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), GST_BUFFER_TIMESTAMP (buffer)); return ret; /* TODO : would be nice if we had some property that told the payloader to put * just 1 frame per RTP packet, for the moment we can set the ptime to 0 or * something smaller or equal to a frame duration */ /* max number of bytes based on given ptime, has to be multiple of * frame_duration */ if (basepayload->max_ptime != -1) { guint ptime_ms = basepayload->max_ptime / 1000000; maxptime_octets = frame_size * (int) (ptime_ms / frame_duration); if (maxptime_octets == 0) { GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %d is smaller than minimum %d ms, overwriting to minimum", ptime_ms, frame_duration); maxptime_octets = frame_size; } /* if the adapter is empty (should be), let's set the base timestamp */ if (gst_adapter_available (basertpaudiopayload->adapter) == 0) { basertpaudiopayload->adapter_base_ts = GST_BUFFER_TIMESTAMP (buffer); "Adapter should be empty but is not!"); gst_adapter_push (basertpaudiopayload->adapter, buffer); available = gst_adapter_available (basertpaudiopayload->adapter); /* as long as we have full frames */ /* this loop will always empty the adapter till the last frame */ /* TODO Make it possible to set a minimum size per packet, this way the * algorithm doesn't empty the adapter if there is too little data left and * will wait until the next buffers to arrive */ while (available >= frame_size) { /* we need to see how many frames we can get based on maximum MTU, maximum * ptime and the number of bytes available in the adapter */ payload_len = MIN (MIN ( /* MTU max */ (int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU (basertpaudiopayload), 0, 0) / frame_size) * frame_size, /* ptime max */ maxptime_octets), /* currently available */ floor (available / frame_size) * frame_size); data = (guint8 *) gst_adapter_peek (basertpaudiopayload->adapter, payload_len); ret = gst_basertpaudiopayload_push (basepayload, data, payload_len, basertpaudiopayload->adapter_base_ts); gst_adapter_flush (basertpaudiopayload->adapter, payload_len); gfloat ts_inc = (payload_len * frame_duration) / frame_size; ts_inc = ts_inc * GST_MSECOND; basertpaudiopayload->adapter_base_ts += ts_inc; GST_DEBUG_OBJECT (basertpaudiopayload, "%f %f %d", ts_inc, ts_inc * GST_MSECOND, (payload_len * frame_duration) / frame_size); GST_DEBUG_OBJECT (basertpaudiopayload, "Pushing with ts %" GST_TIME_FORMAT, GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts)); available = gst_adapter_available (basertpaudiopayload->adapter); /* adapter should be freed by now */ if (available != 0) { guint minptime_octets = 0; guint sample_size; if (basertpaudiopayload->sample_size == 0) { sample_size = basertpaudiopayload->sample_size; /* max number of bytes based on given ptime */ maxptime_octets = basepayload->max_ptime * basepayload->clock_rate / (sample_size * GST_SECOND); minptime_octets = GST_RTP_MIN_PTIME_MS * basepayload->clock_rate / (sample_size * 1000); GST_DEBUG_OBJECT (basertpaudiopayload, "Calculated max_octects %u and min_octets %u", maxptime_octets, minptime_octets); if (maxptime_octets < minptime_octets) { "Given ptime %d is smaller than minimum %d, replacing by %d", maxptime_octets, minptime_octets, minptime_octets); maxptime_octets = minptime_octets; GST_DEBUG_OBJECT (basertpaudiopayload, "Setting to %" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer))); while (available >= minptime_octets) { gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU (basertpaudiopayload), 0, 0), available); gfloat num = payload_len; gfloat datarate = (sample_size * basepayload->clock_rate); basertpaudiopayload->adapter_base_ts += /* payload_len (bytes) * nsecs/sec / datarate (bytes*sec) */ num / datarate * GST_SECOND; GST_DEBUG_OBJECT (basertpaudiopayload, "Calculating ts inc %f %f %f", num, datarate, num / datarate * GST_SECOND); GST_DEBUG_OBJECT (basertpaudiopayload, "New ts is %" GST_TIME_FORMAT, guint payload_len, GstClockTime timestamp) GstBuffer *outbuf; guint8 *payload; /* create buffer to hold the payload */ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); /* copy payload */ gst_rtp_buffer_set_payload_type (outbuf, basepayload->pt); payload = gst_rtp_buffer_get_payload (outbuf); memcpy (payload, data, payload_len); GST_BUFFER_TIMESTAMP (outbuf) = timestamp; ret = gst_basertppayload_push (basepayload, outbuf); --- NEW FILE: gstbasertpaudiopayload.h --- #ifndef __GST_BASE_RTP_AUDIO_PAYLOAD_H__ #define __GST_BASE_RTP_AUDIO_PAYLOAD_H__ #include <gst/gst.h> #include <gst/rtp/gstbasertppayload.h> #include <gst/base/gstadapter.h> G_BEGIN_DECLS typedef struct _GstBaseRTPAudioPayload GstBaseRTPAudioPayload; typedef struct _GstBaseRTPAudioPayloadClass GstBaseRTPAudioPayloadClass; #define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \ (gst_basertpaudiopayload_get_type()) #define GST_BASE_RTP_AUDIO_PAYLOAD(obj) \ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload)) #define GST_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload)) #define GST_IS_BASE_RTP_AUDIO_PAYLOAD(obj) \ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD)) #define GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS(obj) \ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD)) typedef enum { AUDIO_CODEC_TYPE_NONE, AUDIO_CODEC_TYPE_FRAME_BASED, AUDIO_CODEC_TYPE_SAMPLE_BASED } AudioCodecType; struct _GstBaseRTPAudioPayload GstBaseRTPPayload payload; GstClockTime adapter_base_ts; GstAdapter *adapter; gint frame_size; gint frame_duration; gint sample_size; AudioCodecType type; }; struct _GstBaseRTPAudioPayloadClass GstBaseRTPPayloadClass parent_class; gboolean gst_basertpaudiopayload_plugin_init (GstPlugin * plugin); GType gst_basertpaudiopayload_get_type (void); gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload); gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload); *basertpaudiopayload, gint frame_duration, gint frame_size); *basertpaudiopayload, gint sample_size); G_END_DECLS #endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */ Index: gstrtpbuffer.h RCS file: /cvs/gstreamer/gst-plugins-base/gst-libs/gst/rtp/gstrtpbuffer.h,v retrieving revision 1.8 retrieving revision 1.9 diff -u -d -r1.8 -r1.9 --- gstrtpbuffer.h 6 Dec 2005 19:42:01 -0000 1.8 +++ gstrtpbuffer.h 13 Apr 2006 03:55:12 -0000 1.9 @@ -71,6 +71,8 @@ #define GST_RTP_PAYLOAD_MPV_STRING "32" #define GST_RTP_PAYLOAD_H263_STRING "34" +#define GST_RTP_PAYLOAD_DYNAMIC_STRING "[96, 127]" /* creating buffers */ GstBuffer* gst_rtp_buffer_new (void); void gst_rtp_buffer_allocate_data (GstBuffer *buffer, guint payload_len, |