*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/audio/audio.h>
#include "gstrtpopuspay.h"
+#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
#define GST_CAT_DEFAULT (rtpopuspay_debug)
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-opus")
+ GST_STATIC_CAPS ("audio/x-opus, channel-mapping-family = (int) 0")
);
static GstStaticPadTemplate gst_rtp_opus_pay_src_template =
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 48000, "
- "encoding-name = (string) \"X-GST-OPUS-DRAFT-SPITTKA-00\"")
+ "encoding-params = (string) \"2\", "
+ "encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\" }")
);
-static gboolean gst_rtp_opus_pay_setcaps (GstBaseRTPPayload * payload,
+static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
-static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstBaseRTPPayload *
+static GstCaps *gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
+ GstPad * pad, GstCaps * filter);
+static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
-GST_BOILERPLATE (GstRtpOPUSPay, gst_rtp_opus_pay, GstBaseRTPPayload,
- GST_TYPE_BASE_RTP_PAYLOAD);
+G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
-gst_rtp_opus_pay_base_init (gpointer klass)
+gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstRTPBasePayloadClass *gstbasertppayload_class;
+ GstElementClass *element_class;
+
+ gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
+ element_class = GST_ELEMENT_CLASS (klass);
+
+ gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
+ gstbasertppayload_class->get_caps = gst_rtp_opus_pay_getcaps;
+ gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_opus_pay_src_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_opus_pay_sink_template));
+ gst_element_class_add_static_pad_template (element_class,
+ &gst_rtp_opus_pay_src_template);
+ gst_element_class_add_static_pad_template (element_class,
+ &gst_rtp_opus_pay_sink_template);
- gst_element_class_set_details_simple (element_class,
+ gst_element_class_set_static_metadata (element_class,
"RTP Opus payloader",
"Codec/Payloader/Network/RTP",
"Puts Opus audio in RTP packets",
"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
-}
-
-static void
-gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
-{
- GstBaseRTPPayloadClass *gstbasertppayload_class;
-
- gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
-
- gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
- gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
"Opus RTP Payloader");
}
static void
-gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay, GstRtpOPUSPayClass * klass)
+gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
{
}
static gboolean
-gst_rtp_opus_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
- gchar *capsstr;
+ GstCaps *src_caps;
+ GstStructure *s;
+ const char *encoding_name = "OPUS";
+ gint channels, rate;
+ const char *sprop_stereo = NULL;
+ char *sprop_maxcapturerate = NULL;
+
+ src_caps = gst_pad_get_allowed_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
+ if (src_caps) {
+ GstStructure *s;
+ const GValue *value;
+
+ s = gst_caps_get_structure (src_caps, 0);
+
+ if (gst_structure_has_field (s, "encoding-name")) {
+ GValue default_value = G_VALUE_INIT;
+
+ g_value_init (&default_value, G_TYPE_STRING);
+ g_value_set_static_string (&default_value, encoding_name);
+
+ value = gst_structure_get_value (s, "encoding-name");
+ if (!gst_value_can_intersect (&default_value, value))
+ encoding_name = "X-GST-OPUS-DRAFT-SPITTKA-00";
+ }
+ gst_caps_unref (src_caps);
+ }
- capsstr = gst_caps_to_string (caps);
+ s = gst_caps_get_structure (caps, 0);
+ if (gst_structure_get_int (s, "channels", &channels)) {
+ if (channels > 2) {
+ GST_ERROR_OBJECT (payload,
+ "More than 2 channels with channel-mapping-family=0 is invalid");
+ return FALSE;
+ } else if (channels == 2) {
+ sprop_stereo = "1";
+ } else {
+ sprop_stereo = "0";
+ }
+ }
- gst_basertppayload_set_options (payload, "audio", FALSE,
- "X-GST-OPUS-DRAFT-SPITTKA-00", 48000);
- res =
- gst_basertppayload_set_outcaps (payload, "caps", G_TYPE_STRING, capsstr,
- NULL);
- g_free (capsstr);
+ if (gst_structure_get_int (s, "rate", &rate)) {
+ sprop_maxcapturerate = g_strdup_printf ("%d", rate);
+ }
+
+ gst_rtp_base_payload_set_options (payload, "audio", FALSE,
+ encoding_name, 48000);
+
+ if (sprop_maxcapturerate && sprop_stereo) {
+ res =
+ gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
+ G_TYPE_STRING, sprop_maxcapturerate, "sprop-stereo", G_TYPE_STRING,
+ sprop_stereo, NULL);
+ } else if (sprop_maxcapturerate) {
+ res =
+ gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
+ G_TYPE_STRING, sprop_maxcapturerate, NULL);
+ } else if (sprop_stereo) {
+ res =
+ gst_rtp_base_payload_set_outcaps (payload, "sprop-stereo",
+ G_TYPE_STRING, sprop_stereo, NULL);
+ } else {
+ res = gst_rtp_base_payload_set_outcaps (payload, NULL);
+ }
+
+ g_free (sprop_maxcapturerate);
return res;
}
static GstFlowReturn
-gst_rtp_opus_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstBuffer *outbuf;
- GstClockTime timestamp;
+ GstClockTime pts, dts, duration;
+
+ pts = GST_BUFFER_PTS (buffer);
+ dts = GST_BUFFER_DTS (buffer);
+ duration = GST_BUFFER_DURATION (buffer);
+
+ outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
- guint size;
- guint8 *data;
- guint8 *payload;
+ gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
- size = GST_BUFFER_SIZE (buffer);
- data = GST_BUFFER_DATA (buffer);
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ outbuf = gst_buffer_append (outbuf, buffer);
+
+ GST_BUFFER_PTS (outbuf) = pts;
+ GST_BUFFER_DTS (outbuf) = dts;
+ GST_BUFFER_DURATION (outbuf) = duration;
+
+ /* Push out */
+ return gst_rtp_base_payload_push (basepayload, outbuf);
+}
+
+static GstCaps *
+gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
+ GstPad * pad, GstCaps * filter)
+{
+ GstCaps *caps, *peercaps, *tcaps;
+ GstStructure *s;
+ const gchar *stereo;
- outbuf = gst_rtp_buffer_new_allocate (size, 0, 0);
- payload = gst_rtp_buffer_get_payload (outbuf);
+ if (pad == GST_RTP_BASE_PAYLOAD_SRCPAD (payload))
+ return
+ GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
+ (payload, pad, filter);
- memcpy (payload, data, size);
+ tcaps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
+ peercaps = gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload),
+ tcaps);
+ gst_caps_unref (tcaps);
+ if (!peercaps)
+ return
+ GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
+ (payload, pad, filter);
- gst_rtp_buffer_set_marker (outbuf, FALSE);
- GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
+ if (gst_caps_is_empty (peercaps))
+ return peercaps;
- return gst_basertppayload_push (basepayload, outbuf);
+ caps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SINKPAD (payload));
+
+ s = gst_caps_get_structure (peercaps, 0);
+ stereo = gst_structure_get_string (s, "stereo");
+ if (stereo != NULL) {
+ caps = gst_caps_make_writable (caps);
+
+ if (!strcmp (stereo, "1")) {
+ GstCaps *caps2 = gst_caps_copy (caps);
+
+ gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
+ gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 1, NULL);
+ caps = gst_caps_merge (caps, caps2);
+ } else if (!strcmp (stereo, "0")) {
+ GstCaps *caps2 = gst_caps_copy (caps);
+
+ gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
+ gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 2, NULL);
+ caps = gst_caps_merge (caps, caps2);
+ }
+ }
+ gst_caps_unref (peercaps);
+
+ if (filter) {
+ GstCaps *tmp = gst_caps_intersect_full (caps, filter,
+ GST_CAPS_INTERSECT_FIRST);
+ gst_caps_unref (caps);
+ caps = tmp;
+ }
+
+ GST_DEBUG_OBJECT (payload, "Returning caps: %" GST_PTR_FORMAT, caps);
+ return caps;
+}
+
+gboolean
+gst_rtp_opus_pay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpopuspay",
+ GST_RANK_PRIMARY, GST_TYPE_RTP_OPUS_PAY);
}
projects / platform / upstream / gst-plugins-good.git / blobdiff