*
* 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)
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
- "format = (string) S16_BE, "
+ "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" */
);
#define gst_rtp_L16_depay_parent_class parent_class
-G_DEFINE_TYPE (GstRtpL16Depay, gst_rtp_L16_depay, GST_TYPE_BASE_RTP_DEPAYLOAD);
+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 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;
- gstbasertpdepayload_class->set_caps = gst_rtp_L16_depay_setcaps;
- gstbasertpdepayload_class->process = gst_rtp_L16_depay_process;
+ gstrtpbasedepayload_class->set_caps = gst_rtp_L16_depay_setcaps;
+ gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_L16_depay_process;
- 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));
+ 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);
- gst_element_class_set_details_simple (gstelement_class,
+ 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>");
static void
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);
}
depayload->clock_rate = clock_rate;
- rtpL16depay->rate = clock_rate;
- rtpL16depay->channels = channels;
- srccaps = gst_caps_new_simple ("audio/x-raw",
- "format", G_TYPE_STRING, "S16_BE",
- "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);
}
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;
- GstRTPBuffer rtp;
+ GstAudioInfo *info;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
- gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
- payload_len = gst_rtp_buffer_get_payload_len (&rtp);
+ 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 (&rtp);
- marker = gst_rtp_buffer_get_marker (&rtp);
+ 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);
}
- gst_rtp_buffer_unmap (&rtp);
+ 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;
{
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
("Empty Payload."), (NULL));
- gst_rtp_buffer_unmap (&rtp);
+ return NULL;
+ }
+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;
}
}
projects / platform / upstream / gst-plugins-good.git / blobdiff