*
* 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.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:element-rtpL16depay
+ * @see_also: rtpL16pay
+ *
+ * Extract raw audio from RTP packets according to RFC 3551.
+ * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
+ *
+ * <refsect2>
+ * <title>Example pipeline</title>
+ * |[
+ * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink
+ * ]| This example pipeline will depayload an RTP raw audio stream. Refer to
+ * the rtpL16pay example to create the RTP stream.
+ * </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include <stdlib.h>
#include <gst/audio/audio.h>
-#include <gst/audio/multichannel.h>
#include "gstrtpL16depay.h"
#include "gstrtpchannels.h"
+#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpL16depay_debug);
#define GST_CAT_DEFAULT (rtpL16depay_debug)
-/* elementfactory information */
-static const GstElementDetails gst_rtp_L16_depay_details =
-GST_ELEMENT_DETAILS ("RTP audio depayloader",
- "Codec/Depayloader/Network",
- "Extracts raw audio from RTP packets",
- "Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>");
-
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) BIG_ENDIAN, "
- "signed = (boolean) true, "
- "width = (int) 16, "
- "depth = (int) 16, "
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) S16BE, "
+ "layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
- "media = (string) \"audio\", "
- "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
- "clock-rate = (int) [ 1, MAX ], "
+ "media = (string) \"audio\", " "clock-rate = (int) [ 1, MAX ], "
/* "channels = (int) [1, MAX]" */
/* "emphasis = (string) ANY" */
/* "channel-order = (string) ANY" */
)
);
-GST_BOILERPLATE (GstRtpL16Depay, gst_rtp_L16_depay, GstBaseRTPDepayload,
- GST_TYPE_BASE_RTP_DEPAYLOAD);
+#define gst_rtp_L16_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpL16Depay, gst_rtp_L16_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
-static gboolean gst_rtp_L16_depay_setcaps (GstBaseRTPDepayload * depayload,
+static gboolean gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
-static GstBuffer *gst_rtp_L16_depay_process (GstBaseRTPDepayload * depayload,
- GstBuffer * buf);
-
-static GstStateChangeReturn gst_rtp_L16_depay_change_state (GstElement *
- element, GstStateChange transition);
-
-static void
-gst_rtp_L16_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_L16_depay_src_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_L16_depay_sink_template));
-
- gst_element_class_set_details (element_class, &gst_rtp_L16_depay_details);
-}
+static GstBuffer *gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload,
+ GstRTPBuffer * rtp);
static void
gst_rtp_L16_depay_class_init (GstRtpL16DepayClass * klass)
{
GstElementClass *gstelement_class;
- GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+ GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gstelement_class = (GstElementClass *) klass;
- gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+ gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
- parent_class = g_type_class_peek_parent (klass);
+ gstrtpbasedepayload_class->set_caps = gst_rtp_L16_depay_setcaps;
+ gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_L16_depay_process;
- gstelement_class->change_state = gst_rtp_L16_depay_change_state;
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_L16_depay_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_L16_depay_sink_template);
- gstbasertpdepayload_class->set_caps = gst_rtp_L16_depay_setcaps;
- gstbasertpdepayload_class->process = gst_rtp_L16_depay_process;
+ 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>");
GST_DEBUG_CATEGORY_INIT (rtpL16depay_debug, "rtpL16depay", 0,
"Raw Audio RTP Depayloader");
}
static void
-gst_rtp_L16_depay_init (GstRtpL16Depay * rtpL16depay,
- GstRtpL16DepayClass * klass)
+gst_rtp_L16_depay_init (GstRtpL16Depay * rtpL16depay)
{
- /* needed because of GST_BOILERPLATE */
}
static gint
}
static gboolean
-gst_rtp_L16_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpL16Depay *rtpL16depay;
gboolean res;
const gchar *channel_order;
const GstRTPChannelOrder *order;
+ GstAudioInfo *info;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
clock_rate = 44100;
break;
default:
- /* no fixed mapping, we need channels and clock-rate */
+ /* no fixed mapping, we need clock-rate */
channels = 0;
clock_rate = 0;
break;
if (clock_rate == 0)
goto no_clockrate;
- channels = gst_rtp_L16_depay_parse_int (structure, "channels", channels);
- if (channels == 0)
- goto no_channels;
+ 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;
+ }
+ }
depayload->clock_rate = clock_rate;
- rtpL16depay->rate = clock_rate;
- rtpL16depay->channels = channels;
- srccaps = gst_caps_new_simple ("audio/x-raw-int",
- "endianness", G_TYPE_INT, G_BIG_ENDIAN,
- "signed", G_TYPE_BOOLEAN, TRUE,
- "width", G_TYPE_INT, 16,
- "depth", G_TYPE_INT, 16,
- "rate", G_TYPE_INT, clock_rate, "channels", G_TYPE_INT, channels, NULL);
+ 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) {
- gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
- order->pos);
+ memcpy (info->position, order->pos,
+ sizeof (GstAudioChannelPosition) * channels);
+ gst_audio_channel_positions_to_valid_order (info->position, info->channels);
} else {
- GstAudioChannelPosition *pos;
-
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
(NULL), ("Unknown channel order '%s' for %d channels",
GST_STR_NULL (channel_order), channels));
/* create default NONE layout */
- pos = gst_rtp_channels_create_default (channels);
- gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
- g_free (pos);
+ gst_rtp_channels_create_default (channels, info->position);
}
+ srccaps = gst_audio_info_to_caps (info);
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
GST_ERROR_OBJECT (depayload, "no clock-rate specified");
return FALSE;
}
-no_channels:
- {
- GST_ERROR_OBJECT (depayload, "no channels specified");
- return FALSE;
- }
}
static GstBuffer *
-gst_rtp_L16_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpL16Depay *rtpL16depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
+ GstAudioInfo *info;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
- payload_len = gst_rtp_buffer_get_payload_len (buf);
+ payload_len = gst_rtp_buffer_get_payload_len (rtp);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
- outbuf = gst_rtp_buffer_get_payload_buffer (buf);
- marker = gst_rtp_buffer_get_marker (buf);
+ outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
+ marker = gst_rtp_buffer_get_marker (rtp);
if (marker) {
- /* mark talk spurt with DISCONT */
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ /* mark talk spurt with RESYNC */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
+ }
+
+ outbuf = gst_buffer_make_writable (outbuf);
+ info = &rtpL16depay->info;
+
+ if (payload_len % info->bpf != 0)
+ goto wrong_payload_size;
+
+ if (rtpL16depay->order &&
+ !gst_audio_buffer_reorder_channels (outbuf,
+ info->finfo->format, info->channels,
+ info->position, rtpL16depay->order->pos)) {
+ goto reorder_failed;
}
+ gst_rtp_drop_non_audio_meta (rtpL16depay, outbuf);
+
return outbuf;
/* ERRORS */
("Empty Payload."), (NULL));
return NULL;
}
-}
-
-static GstStateChangeReturn
-gst_rtp_L16_depay_change_state (GstElement * element, GstStateChange transition)
-{
- GstRtpL16Depay *rtpL16depay;
- GstStateChangeReturn ret;
-
- rtpL16depay = GST_RTP_L16_DEPAY (element);
-
- /*
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- break;
- default:
- break;
- }
- */
-
- ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
- /*
- switch (transition) {
- case GST_STATE_CHANGE_READY_TO_NULL:
- break;
- default:
- break;
- }
- */
- return ret;
+wrong_payload_size:
+ {
+ GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
+ ("Wrong Payload Size."), (NULL));
+ return NULL;
+ }
+reorder_failed:
+ {
+ GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
+ ("Channel reordering failed."), (NULL));
+ return NULL;
+ }
}
gboolean
gst_rtp_L16_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpL16depay",
- GST_RANK_MARGINAL, GST_TYPE_RTP_L16_DEPAY);
+ GST_RANK_SECONDARY, GST_TYPE_RTP_L16_DEPAY);
}
projects / platform / upstream / gst-plugins-good.git / blobdiff