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gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
Original commit message from CVS: 2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk> * 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
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@ -1,3 +1,11 @@
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2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
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* gst-libs/gst/rtp/gstrtpbuffer.h:
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Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
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New RTP audio base payloader class. Supports frame or sample based codecs
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2006-04-12 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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@ -2,12 +2,14 @@ libgstrtpincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/rtp
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libgstrtpinclude_HEADERS = gstrtpbuffer.h \
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gstbasertppayload.h \
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gstbasertpaudiopayload.h \
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gstbasertpdepayload.h
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lib_LTLIBRARIES = libgstrtp-@GST_MAJORMINOR@.la
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libgstrtp_@GST_MAJORMINOR@_la_SOURCES = gstrtpbuffer.c \
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gstbasertppayload.c \
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gstbasertpaudiopayload.c \
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gstbasertpdepayload.c
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libgstrtp_@GST_MAJORMINOR@_la_CFLAGS = $(GST_CFLAGS)
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429
gst-libs/gst/rtp/gstbasertpaudiopayload.c
Normal file
429
gst-libs/gst/rtp/gstbasertpaudiopayload.c
Normal file
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/* GStreamer
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* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <math.h>
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#include "gstbasertpaudiopayload.h"
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GST_DEBUG_CATEGORY (basertpaudiopayload_debug);
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#define GST_CAT_DEFAULT (basertpaudiopayload_debug)
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/* let us define a minimum of 10 ms for sample based codecs */
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#define GST_RTP_MIN_PTIME_MS 10
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static void gst_basertpaudiopayload_finalize (GObject * object);
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static GstFlowReturn
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gst_basertpaudiopayload_push (GstBaseRTPPayload * basepayload, guint8 * data,
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guint payload_len, GstClockTime timestamp);
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static GstFlowReturn gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buffer);
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static GstFlowReturn
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gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload *
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basepayload, GstBuffer * buffer);
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static GstFlowReturn
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gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload *
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basepayload, GstBuffer * buffer);
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GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_basertpaudiopayload,
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GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD);
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static void
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gst_basertpaudiopayload_base_init (gpointer klass)
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{
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}
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static void
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gst_basertpaudiopayload_class_init (GstBaseRTPAudioPayloadClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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gobject_class->finalize = gst_basertpaudiopayload_finalize;
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parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
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gstbasertppayload_class->handle_buffer =
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gst_basertpaudiopayload_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0,
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"base audio RTP payloader");
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}
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static void
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gst_basertpaudiopayload_init (GstBaseRTPAudioPayload * basertpaudiopayload,
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GstBaseRTPAudioPayloadClass * klass)
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{
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basertpaudiopayload->adapter = gst_adapter_new ();
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basertpaudiopayload->adapter_base_ts = 0;
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basertpaudiopayload->type = AUDIO_CODEC_TYPE_NONE;
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/* these need to be set by child object if frame based */
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basertpaudiopayload->frame_size = 0;
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basertpaudiopayload->frame_duration = 0;
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/* these need to be set by child object if sample based */
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basertpaudiopayload->sample_size = 0;
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}
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static void
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gst_basertpaudiopayload_finalize (GObject * object)
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{
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
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g_object_unref (basertpaudiopayload->adapter);
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basertpaudiopayload->adapter = NULL;
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GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
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}
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void
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gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload *
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basertpaudiopayload)
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{
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g_return_if_fail (basertpaudiopayload != NULL);
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if (basertpaudiopayload->type != AUDIO_CODEC_TYPE_NONE) {
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GST_ERROR_OBJECT (basertpaudiopayload,
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"Codec type already set! You should only set this once!");
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}
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basertpaudiopayload->type = AUDIO_CODEC_TYPE_FRAME_BASED;
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}
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void
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gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload *
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basertpaudiopayload)
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{
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g_return_if_fail (basertpaudiopayload != NULL);
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if (basertpaudiopayload->type != AUDIO_CODEC_TYPE_NONE) {
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GST_ERROR_OBJECT (basertpaudiopayload,
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"Codec type already set! You should only set this once!");
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}
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basertpaudiopayload->type = AUDIO_CODEC_TYPE_SAMPLE_BASED;
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}
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/* These are options that need to be set for frame based audio codecs */
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void
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gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
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* basertpaudiopayload, gint frame_duration, gint frame_size)
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{
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g_return_if_fail (basertpaudiopayload != NULL);
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basertpaudiopayload->frame_size = frame_size;
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basertpaudiopayload->frame_duration = frame_duration;
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}
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void
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gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload
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* basertpaudiopayload, gint sample_size)
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{
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g_return_if_fail (basertpaudiopayload != NULL);
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basertpaudiopayload->sample_size = sample_size;
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}
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static GstFlowReturn
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gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstFlowReturn ret;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
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ret = GST_FLOW_ERROR;
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if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_FRAME_BASED) {
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ret = gst_basertpaudiopayload_handle_frame_based_buffer (basepayload,
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buffer);
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} else if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) {
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ret = gst_basertpaudiopayload_handle_sample_based_buffer (basepayload,
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buffer);
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} else {
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GST_DEBUG_OBJECT (basertpaudiopayload, "Audio codec type not set");
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}
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return ret;
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}
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/* this assumes all frames have a constant duration and a constant size */
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static GstFlowReturn
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gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload *
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basepayload, GstBuffer * buffer)
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{
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GstBaseRTPAudioPayload *basertpaudiopayload;
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guint payload_len;
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guint8 *data;
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GstFlowReturn ret;
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guint available;
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gint frame_size, frame_duration;
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guint maxptime_octets = G_MAXUINT;
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ret = GST_FLOW_ERROR;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
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if (basertpaudiopayload->frame_size == 0 ||
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basertpaudiopayload->frame_duration == 0) {
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GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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frame_size = basertpaudiopayload->frame_size;
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frame_duration = basertpaudiopayload->frame_duration;
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/* If buffer fits on an RTP packet, let's just push it through without using
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* the adapter */
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/* this will check again max_ptime and max_mtu */
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if (!gst_basertppayload_is_filled (basepayload,
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gst_rtp_buffer_calc_packet_len (GST_BUFFER_SIZE (buffer), 0, 0),
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GST_BUFFER_DURATION (buffer))) {
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ret = gst_basertpaudiopayload_push (basepayload, GST_BUFFER_DATA (buffer),
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GST_BUFFER_SIZE (buffer), GST_BUFFER_TIMESTAMP (buffer));
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gst_buffer_unref (buffer);
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return ret;
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}
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/* TODO : would be nice if we had some property that told the payloader to put
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* just 1 frame per RTP packet, for the moment we can set the ptime to 0 or
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* something smaller or equal to a frame duration */
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/* max number of bytes based on given ptime, has to be multiple of
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* frame_duration */
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if (basepayload->max_ptime != -1) {
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guint ptime_ms = basepayload->max_ptime / 1000000;
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maxptime_octets = frame_size * (int) (ptime_ms / frame_duration);
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if (maxptime_octets == 0) {
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GST_WARNING_OBJECT (basertpaudiopayload,
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"Given ptime %d is smaller than minimum %d ms, overwriting to minimum",
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ptime_ms, frame_duration);
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maxptime_octets = frame_size;
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}
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}
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/* if the adapter is empty (should be), let's set the base timestamp */
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if (gst_adapter_available (basertpaudiopayload->adapter) == 0) {
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basertpaudiopayload->adapter_base_ts = GST_BUFFER_TIMESTAMP (buffer);
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} else {
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GST_ERROR_OBJECT (basertpaudiopayload,
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"Adapter should be empty but is not!");
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return GST_FLOW_ERROR;
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}
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gst_adapter_push (basertpaudiopayload->adapter, buffer);
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available = gst_adapter_available (basertpaudiopayload->adapter);
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/* as long as we have full frames */
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/* this loop will always empty the adapter till the last frame */
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/* TODO Make it possible to set a minimum size per packet, this way the
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* algorithm doesn't empty the adapter if there is too little data left and
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* will wait until the next buffers to arrive */
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while (available >= frame_size) {
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/* we need to see how many frames we can get based on maximum MTU, maximum
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* ptime and the number of bytes available in the adapter */
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payload_len = MIN (MIN (
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/* MTU max */
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(int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
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(basertpaudiopayload), 0, 0) / frame_size) * frame_size,
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/* ptime max */
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maxptime_octets),
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/* currently available */
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floor (available / frame_size) * frame_size);
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data =
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(guint8 *) gst_adapter_peek (basertpaudiopayload->adapter, payload_len);
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ret =
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gst_basertpaudiopayload_push (basepayload, data, payload_len,
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basertpaudiopayload->adapter_base_ts);
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gst_adapter_flush (basertpaudiopayload->adapter, payload_len);
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gfloat ts_inc = (payload_len * frame_duration) / frame_size;
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ts_inc = ts_inc * GST_MSECOND;
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basertpaudiopayload->adapter_base_ts += ts_inc;
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GST_DEBUG_OBJECT (basertpaudiopayload, "%f %f %d", ts_inc,
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ts_inc * GST_MSECOND, (payload_len * frame_duration) / frame_size);
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GST_DEBUG_OBJECT (basertpaudiopayload, "Pushing with ts %" GST_TIME_FORMAT,
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GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts));
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available = gst_adapter_available (basertpaudiopayload->adapter);
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}
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/* adapter should be freed by now */
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if (available != 0) {
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GST_ERROR_OBJECT (basertpaudiopayload,
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"Adapter should be empty but is not!");
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return GST_FLOW_ERROR;
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}
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return ret;
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}
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static GstFlowReturn
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gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload *
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basepayload, GstBuffer * buffer)
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{
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GstBaseRTPAudioPayload *basertpaudiopayload;
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guint payload_len;
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guint8 *data;
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GstFlowReturn ret;
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guint available;
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guint maxptime_octets = G_MAXUINT;
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guint minptime_octets = 0;
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guint sample_size;
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ret = GST_FLOW_ERROR;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
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if (basertpaudiopayload->sample_size == 0) {
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GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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sample_size = basertpaudiopayload->sample_size;
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/* If buffer fits on an RTP packet, let's just push it through without using
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* the adapter */
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/* this will check again max_ptime and max_mtu */
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if (!gst_basertppayload_is_filled (basepayload,
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gst_rtp_buffer_calc_packet_len (GST_BUFFER_SIZE (buffer), 0, 0),
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GST_BUFFER_DURATION (buffer))) {
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ret = gst_basertpaudiopayload_push (basepayload, GST_BUFFER_DATA (buffer),
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GST_BUFFER_SIZE (buffer), GST_BUFFER_TIMESTAMP (buffer));
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gst_buffer_unref (buffer);
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return ret;
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}
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/* max number of bytes based on given ptime */
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if (basepayload->max_ptime != -1) {
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maxptime_octets = basepayload->max_ptime * basepayload->clock_rate /
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(sample_size * GST_SECOND);
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minptime_octets = GST_RTP_MIN_PTIME_MS * basepayload->clock_rate /
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(sample_size * 1000);
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GST_DEBUG_OBJECT (basertpaudiopayload,
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"Calculated max_octects %u and min_octets %u", maxptime_octets,
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minptime_octets);
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if (maxptime_octets < minptime_octets) {
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GST_WARNING_OBJECT (basertpaudiopayload,
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"Given ptime %d is smaller than minimum %d, replacing by %d",
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maxptime_octets, minptime_octets, minptime_octets);
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maxptime_octets = minptime_octets;
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}
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}
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/* if the adapter is empty (should be), let's set the base timestamp */
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if (gst_adapter_available (basertpaudiopayload->adapter) == 0) {
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basertpaudiopayload->adapter_base_ts = GST_BUFFER_TIMESTAMP (buffer);
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GST_DEBUG_OBJECT (basertpaudiopayload, "Setting to %" GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
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}
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gst_adapter_push (basertpaudiopayload->adapter, buffer);
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available = gst_adapter_available (basertpaudiopayload->adapter);
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/* as long as we have full frames */
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/* this loop will always empty the adapter till the last frame */
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/* TODO Make it possible to set a minimum size per packet, this way the
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* algorithm doesn't empty the adapter if there is too little data left and
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* will wait until the next buffers to arrive */
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while (available >= minptime_octets) {
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/* we need to see how many frames we can get based on maximum MTU, maximum
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* ptime and the number of bytes available in the adapter */
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payload_len = MIN (MIN (
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/* MTU max */
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gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
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(basertpaudiopayload), 0, 0),
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/* ptime max */
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||||
maxptime_octets),
|
||||
/* currently available */
|
||||
available);
|
||||
|
||||
data =
|
||||
(guint8 *) gst_adapter_peek (basertpaudiopayload->adapter, payload_len);
|
||||
GST_DEBUG_OBJECT (basertpaudiopayload, "Pushing with ts %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts));
|
||||
ret =
|
||||
gst_basertpaudiopayload_push (basepayload, data, payload_len,
|
||||
basertpaudiopayload->adapter_base_ts);
|
||||
|
||||
gst_adapter_flush (basertpaudiopayload->adapter, payload_len);
|
||||
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,
|
||||
GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts));
|
||||
|
||||
available = gst_adapter_available (basertpaudiopayload->adapter);
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
gst_basertpaudiopayload_push (GstBaseRTPPayload * basepayload, guint8 * data,
|
||||
guint payload_len, GstClockTime timestamp)
|
||||
{
|
||||
GstBuffer *outbuf;
|
||||
guint8 *payload;
|
||||
GstFlowReturn ret;
|
||||
|
||||
/* 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);
|
||||
|
||||
return ret;
|
||||
}
|
88
gst-libs/gst/rtp/gstbasertpaudiopayload.h
Normal file
88
gst-libs/gst/rtp/gstbasertpaudiopayload.h
Normal file
|
@ -0,0 +1,88 @@
|
|||
/* GStreamer
|
||||
* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
|
||||
*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#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);
|
||||
|
||||
void
|
||||
gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload);
|
||||
|
||||
void
|
||||
gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload);
|
||||
|
||||
void
|
||||
gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
|
||||
*basertpaudiopayload, gint frame_duration, gint frame_size);
|
||||
|
||||
void
|
||||
gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload
|
||||
*basertpaudiopayload, gint sample_size);
|
||||
|
||||
G_END_DECLS
|
||||
|
||||
#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */
|
|
@ -71,6 +71,8 @@ typedef enum
|
|||
#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,
|
||||
|
|
Loading…
Reference in a new issue