/* GStreamer
- * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* 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
+ * 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 <string.h>
-#include "gstrtpL16parse.h"
-#include "gstrtp-common.h"
-
-/* elementfactory information */
-static GstElementDetails gst_rtp_L16parse_details = {
- "RTP packet parser",
- "Codec/Parser/Network",
- "Extracts raw audio from RTP packets",
- "Zeeshan Ali <zak147@yahoo.com>"
-};
-
-/* RtpL16Parse signals and args */
-enum
-{
- /* FILL ME */
- LAST_SIGNAL
-};
+#include <stdlib.h>
-enum
-{
- ARG_0,
- ARG_FREQUENCY,
- ARG_PAYLOAD_TYPE
-};
+#include <gst/audio/audio.h>
+
+#include "gstrtpL16depay.h"
+#include "gstrtpchannels.h"
-static GstStaticPadTemplate gst_rtp_L16parse_src_template =
+GST_DEBUG_CATEGORY_STATIC (rtpL16depay_debug);
+#define GST_CAT_DEFAULT (rtpL16depay_debug)
+
+static GstStaticPadTemplate gst_rtp_L16_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "endianness = (int) BYTE_ORDER, "
- "signed = (boolean) true, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "rate = (int) [ 1000, 48000 ], " "channels = (int) [ 1, 2 ]")
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) S16BE, "
+ "layout = (string) interleaved, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
-static GstStaticPadTemplate gst_rtp_L16parse_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
+static GstStaticPadTemplate gst_rtp_L16_depay_sink_template =
+ GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("application/x-rtp")
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) [ 1, MAX ], "
+ /* "channels = (int) [1, MAX]" */
+ /* "emphasis = (string) ANY" */
+ /* "channel-order = (string) ANY" */
+ "encoding-name = (string) \"L16\";"
+ "application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) { " GST_RTP_PAYLOAD_L16_STEREO_STRING ", "
+ GST_RTP_PAYLOAD_L16_MONO_STRING " }," "clock-rate = (int) [ 1, MAX ]"
+ /* "channels = (int) [1, MAX]" */
+ /* "emphasis = (string) ANY" */
+ /* "channel-order = (string) ANY" */
+ )
);
-static void gst_rtp_L16parse_class_init (GstRtpL16ParseClass * klass);
-static void gst_rtp_L16parse_base_init (GstRtpL16ParseClass * klass);
-static void gst_rtp_L16parse_init (GstRtpL16Parse * rtpL16parse);
-
-static void gst_rtp_L16parse_chain (GstPad * pad, GstData * _data);
-
-static void gst_rtp_L16parse_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec);
-static void gst_rtp_L16parse_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec);
-static GstStateChangeReturn gst_rtp_L16parse_change_state (GstElement *
- element);
-
-static GstElementClass *parent_class = NULL;
+#define gst_rtp_L16_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpL16Depay, gst_rtp_L16_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
-static GType
-gst_rtp_L16parse_get_type (void)
-{
- static GType rtpL16parse_type = 0;
-
- if (!rtpL16parse_type) {
- static const GTypeInfo rtpL16parse_info = {
- sizeof (GstRtpL16ParseClass),
- (GBaseInitFunc) gst_rtp_L16parse_base_init,
- NULL,
- (GClassInitFunc) gst_rtp_L16parse_class_init,
- NULL,
- NULL,
- sizeof (GstRtpL16Parse),
- 0,
- (GInstanceInitFunc) gst_rtp_L16parse_init,
- };
-
- rtpL16parse_type =
- g_type_register_static (GST_TYPE_ELEMENT, "GstRtpL16Parse",
- &rtpL16parse_info, 0);
- }
- return rtpL16parse_type;
-}
+static gboolean gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload,
+ GstCaps * caps);
+static GstBuffer *gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload,
+ GstBuffer * buf);
static void
-gst_rtp_L16parse_base_init (GstRtpL16ParseClass * klass)
+gst_rtp_L16_depay_class_init (GstRtpL16DepayClass * klass)
{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_L16parse_src_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_L16parse_sink_template));
- gst_element_class_set_details (element_class, &gst_rtp_L16parse_details);
-}
-
-static void
-gst_rtp_L16parse_class_init (GstRtpL16ParseClass * klass)
-{
- GObjectClass *gobject_class;
GstElementClass *gstelement_class;
+ GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
- gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
+ gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
- parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+ gstrtpbasedepayload_class->set_caps = gst_rtp_L16_depay_setcaps;
+ gstrtpbasedepayload_class->process = gst_rtp_L16_depay_process;
- g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_PAYLOAD_TYPE,
- g_param_spec_int ("payload_type", "payload_type", "payload type",
- G_MININT, G_MAXINT, PAYLOAD_L16_STEREO, G_PARAM_READABLE));
- g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FREQUENCY,
- g_param_spec_int ("frequency", "frequency", "frequency",
- G_MININT, G_MAXINT, 44100, G_PARAM_READWRITE));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtp_L16_depay_src_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtp_L16_depay_sink_template));
- gobject_class->set_property = gst_rtp_L16parse_set_property;
- gobject_class->get_property = gst_rtp_L16parse_get_property;
+ gst_element_class_set_static_metadata (gstelement_class,
+ "RTP audio depayloader", "Codec/Depayloader/Network/RTP",
+ "Extracts raw audio from RTP packets",
+ "Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>");
- gstelement_class->change_state = gst_rtp_L16parse_change_state;
+ GST_DEBUG_CATEGORY_INIT (rtpL16depay_debug, "rtpL16depay", 0,
+ "Raw Audio RTP Depayloader");
}
static void
-gst_rtp_L16parse_init (GstRtpL16Parse * rtpL16parse)
+gst_rtp_L16_depay_init (GstRtpL16Depay * rtpL16depay)
{
- rtpL16parse->srcpad =
- gst_pad_new_from_template (gst_static_pad_template_get
- (&gst_rtp_L16parse_src_template), "src");
- rtpL16parse->sinkpad =
- gst_pad_new_from_template (gst_static_pad_template_get
- (&gst_rtp_L16parse_sink_template), "sink");
- gst_element_add_pad (GST_ELEMENT (rtpL16parse), rtpL16parse->srcpad);
- gst_element_add_pad (GST_ELEMENT (rtpL16parse), rtpL16parse->sinkpad);
- gst_pad_set_chain_function (rtpL16parse->sinkpad, gst_rtp_L16parse_chain);
-
- rtpL16parse->frequency = 44100;
- rtpL16parse->channels = 2;
-
- rtpL16parse->payload_type = PAYLOAD_L16_STEREO;
}
-void
-gst_rtp_L16parse_ntohs (GstBuffer * buf)
+static gint
+gst_rtp_L16_depay_parse_int (GstStructure * structure, const gchar * field,
+ gint def)
{
- gint16 *i, *len;
+ const gchar *str;
+ gint res;
- /* FIXME: is this code correct or even sane at all? */
- i = (gint16 *) GST_BUFFER_DATA (buf);
- len = i + GST_BUFFER_SIZE (buf) / sizeof (gint16 *);
+ if ((str = gst_structure_get_string (structure, field)))
+ return atoi (str);
- for (; i < len; i++) {
- *i = g_ntohs (*i);
- }
-}
-
-void
-gst_rtp_L16_caps_nego (GstRtpL16Parse * rtpL16parse)
-{
- GstCaps *caps;
-
- caps =
- gst_caps_copy (gst_static_caps_get (&gst_rtp_L16parse_src_template.
- static_caps));
-
- gst_caps_set_simple (caps,
- "rate", G_TYPE_INT, rtpL16parse->frequency,
- "channel", G_TYPE_INT, rtpL16parse->channels, NULL);
+ if (gst_structure_get_int (structure, field, &res))
+ return res;
- gst_pad_try_set_caps (rtpL16parse->srcpad, caps);
+ return def;
}
-void
-gst_rtp_L16parse_payloadtype_change (GstRtpL16Parse * rtpL16parse,
- rtp_payload_t pt)
+static gboolean
+gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
- rtpL16parse->payload_type = pt;
-
- switch (pt) {
- case PAYLOAD_L16_MONO:
- rtpL16parse->channels = 1;
+ GstStructure *structure;
+ GstRtpL16Depay *rtpL16depay;
+ gint clock_rate, payload;
+ gint channels;
+ GstCaps *srccaps;
+ gboolean res;
+ const gchar *channel_order;
+ const GstRTPChannelOrder *order;
+ GstAudioInfo *info;
+
+ rtpL16depay = GST_RTP_L16_DEPAY (depayload);
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ payload = 96;
+ gst_structure_get_int (structure, "payload", &payload);
+ switch (payload) {
+ case GST_RTP_PAYLOAD_L16_STEREO:
+ channels = 2;
+ clock_rate = 44100;
break;
- case PAYLOAD_L16_STEREO:
- rtpL16parse->channels = 2;
+ case GST_RTP_PAYLOAD_L16_MONO:
+ channels = 1;
+ clock_rate = 44100;
break;
default:
- g_warning ("unknown payload_t %d\n", pt);
+ /* no fixed mapping, we need clock-rate */
+ channels = 0;
+ clock_rate = 0;
+ break;
}
- gst_rtp_L16_caps_nego (rtpL16parse);
-}
-
-static void
-gst_rtp_L16parse_chain (GstPad * pad, GstData * _data)
-{
- GstBuffer *buf = GST_BUFFER (_data);
- GstRtpL16Parse *rtpL16parse;
- GstBuffer *outbuf;
- Rtp_Packet packet;
- rtp_payload_t pt;
-
- g_return_if_fail (pad != NULL);
- g_return_if_fail (GST_IS_PAD (pad));
- g_return_if_fail (buf != NULL);
-
- rtpL16parse = GST_RTP_L16_PARSE (GST_OBJECT_PARENT (pad));
-
- g_return_if_fail (rtpL16parse != NULL);
- g_return_if_fail (GST_IS_RTP_L16_PARSE (rtpL16parse));
-
- if (GST_IS_EVENT (buf)) {
- GstEvent *event = GST_EVENT (buf);
-
- gst_pad_event_default (pad, event);
-
- return;
+ /* caps can overwrite defaults */
+ clock_rate =
+ gst_rtp_L16_depay_parse_int (structure, "clock-rate", clock_rate);
+ if (clock_rate == 0)
+ goto no_clockrate;
+
+ channels =
+ gst_rtp_L16_depay_parse_int (structure, "encoding-params", channels);
+ if (channels == 0) {
+ channels = gst_rtp_L16_depay_parse_int (structure, "channels", channels);
+ if (channels == 0) {
+ /* channels defaults to 1 otherwise */
+ channels = 1;
+ }
}
- if (GST_PAD_CAPS (rtpL16parse->srcpad) == NULL) {
- gst_rtp_L16_caps_nego (rtpL16parse);
+ depayload->clock_rate = clock_rate;
+
+ info = &rtpL16depay->info;
+ gst_audio_info_init (info);
+ info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S16BE);
+ info->rate = clock_rate;
+ info->channels = channels;
+ info->bpf = (info->finfo->width / 8) * channels;
+
+ /* add channel positions */
+ channel_order = gst_structure_get_string (structure, "channel-order");
+
+ order = gst_rtp_channels_get_by_order (channels, channel_order);
+ rtpL16depay->order = order;
+ if (order) {
+ memcpy (info->position, order->pos,
+ sizeof (GstAudioChannelPosition) * channels);
+ gst_audio_channel_positions_to_valid_order (info->position, info->channels);
+ } else {
+ GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
+ (NULL), ("Unknown channel order '%s' for %d channels",
+ GST_STR_NULL (channel_order), channels));
+ /* create default NONE layout */
+ gst_rtp_channels_create_default (channels, info->position);
}
- packet =
- rtp_packet_new_copy_data (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+ srccaps = gst_audio_info_to_caps (info);
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
+ gst_caps_unref (srccaps);
- pt = rtp_packet_get_payload_type (packet);
+ return res;
- if (pt != rtpL16parse->payload_type) {
- gst_rtp_L16parse_payloadtype_change (rtpL16parse, pt);
+ /* ERRORS */
+no_clockrate:
+ {
+ GST_ERROR_OBJECT (depayload, "no clock-rate specified");
+ return FALSE;
}
-
- outbuf = gst_buffer_new ();
- GST_BUFFER_SIZE (outbuf) = rtp_packet_get_payload_len (packet);
- GST_BUFFER_DATA (outbuf) = g_malloc (GST_BUFFER_SIZE (outbuf));
- GST_BUFFER_TIMESTAMP (outbuf) =
- g_ntohl (rtp_packet_get_timestamp (packet)) * GST_SECOND;
-
- memcpy (GST_BUFFER_DATA (outbuf), rtp_packet_get_payload (packet),
- GST_BUFFER_SIZE (outbuf));
-
- GST_DEBUG ("gst_rtp_L16parse_chain: pushing buffer of size %d",
- GST_BUFFER_SIZE (outbuf));
-
- /* FIXME: According to RFC 1890, this is required, right? */
-#if G_BYTE_ORDER == G_LITTLE_ENDIAN
- gst_rtp_L16parse_ntohs (outbuf);
-#endif
-
- gst_pad_push (rtpL16parse->srcpad, GST_DATA (outbuf));
-
- rtp_packet_free (packet);
- gst_buffer_unref (buf);
}
-static void
-gst_rtp_L16parse_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
+static GstBuffer *
+gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
- GstRtpL16Parse *rtpL16parse;
+ GstRtpL16Depay *rtpL16depay;
+ GstBuffer *outbuf;
+ gint payload_len;
+ gboolean marker;
+ GstRTPBuffer rtp = { NULL };
- g_return_if_fail (GST_IS_RTP_L16_PARSE (object));
- rtpL16parse = GST_RTP_L16_PARSE (object);
+ rtpL16depay = GST_RTP_L16_DEPAY (depayload);
- switch (prop_id) {
- case ARG_PAYLOAD_TYPE:
- gst_rtp_L16parse_payloadtype_change (rtpL16parse,
- g_value_get_int (value));
- break;
- case ARG_FREQUENCY:
- rtpL16parse->frequency = g_value_get_int (value);
- break;
- default:
- break;
- }
-}
+ gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+ payload_len = gst_rtp_buffer_get_payload_len (&rtp);
-static void
-gst_rtp_L16parse_get_property (GObject * object, guint prop_id, GValue * value,
- GParamSpec * pspec)
-{
- GstRtpL16Parse *rtpL16parse;
+ if (payload_len <= 0)
+ goto empty_packet;
- g_return_if_fail (GST_IS_RTP_L16_PARSE (object));
- rtpL16parse = GST_RTP_L16_PARSE (object);
+ GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
- switch (prop_id) {
- case ARG_PAYLOAD_TYPE:
- g_value_set_int (value, rtpL16parse->payload_type);
- break;
- case ARG_FREQUENCY:
- g_value_set_int (value, rtpL16parse->frequency);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
+ outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
+ marker = gst_rtp_buffer_get_marker (&rtp);
-static GstStateChangeReturn
-gst_rtp_L16parse_change_state (GstElement * element, GstStateChange transition)
-{
- GstRtpL16Parse *rtpL16parse;
+ if (marker) {
+ /* mark talk spurt with DISCONT */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ }
- g_return_val_if_fail (GST_IS_RTP_L16_PARSE (element),
- GST_STATE_CHANGE_FAILURE);
+ outbuf = gst_buffer_make_writable (outbuf);
+ if (rtpL16depay->order &&
+ !gst_audio_buffer_reorder_channels (outbuf,
+ rtpL16depay->info.finfo->format, rtpL16depay->info.channels,
+ rtpL16depay->info.position, rtpL16depay->order->pos)) {
+ goto reorder_failed;
+ }
- rtpL16parse = GST_RTP_L16_PARSE (element);
+ gst_rtp_buffer_unmap (&rtp);
- GST_DEBUG ("state pending %d\n", GST_STATE_PENDING (element));
+ return outbuf;
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
- break;
- default:
- break;
+ /* ERRORS */
+empty_packet:
+ {
+ GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
+ ("Empty Payload."), (NULL));
+ gst_rtp_buffer_unmap (&rtp);
+ return NULL;
+ }
+reorder_failed:
+ {
+ GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
+ ("Channel reordering failed."), (NULL));
+ gst_rtp_buffer_unmap (&rtp);
+ return NULL;
}
-
- /* if we haven't failed already, give the parent class a chance to ;-) */
- if (GST_ELEMENT_CLASS (parent_class)->change_state)
- return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
- return GST_STATE_CHANGE_SUCCESS;
}
gboolean
-gst_rtp_L16parse_plugin_init (GstPlugin * plugin)
+gst_rtp_L16_depay_plugin_init (GstPlugin * plugin)
{
- return gst_element_register (plugin, "rtpL16parse",
- GST_RANK_NONE, GST_TYPE_RTP_L16_PARSE);
+ return gst_element_register (plugin, "rtpL16depay",
+ GST_RANK_SECONDARY, GST_TYPE_RTP_L16_DEPAY);
}
projects / platform / upstream / gstreamer.git / blobdiff