diff options
Diffstat (limited to 'gst/siren/gstrtpsirenpay.c')
-rw-r--r-- | gst/siren/gstrtpsirenpay.c | 163 |
1 files changed, 163 insertions, 0 deletions
diff --git a/gst/siren/gstrtpsirenpay.c b/gst/siren/gstrtpsirenpay.c new file mode 100644 index 00000000..dff8ae7b --- /dev/null +++ b/gst/siren/gstrtpsirenpay.c @@ -0,0 +1,163 @@ +/* + * Siren Payloader Gst Element + * + * @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net> + * + * 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. + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "gstrtpsirenpay.h" +#include <gst/rtp/gstrtpbuffer.h> + +/* elementfactory information */ +static GstElementDetails gst_rtpsirenpay_details = { + "RTP Payloader for Siren Audio", + "Codec/Payloader/Network", + "Packetize Siren audio streams into RTP packets", + "Youness Alaoui <kakaroto@kakaroto.homelinux.net>" +}; + +GST_DEBUG_CATEGORY_STATIC (rtpsirenpay_debug); +#define GST_CAT_DEFAULT (rtpsirenpay_debug) + +static GstStaticPadTemplate gst_rtpsirenpay_sink_template = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320") + ); + +static GstStaticPadTemplate gst_rtpsirenpay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("application/x-rtp, " + "media = (string) \"audio\", " + "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " + "clock-rate = (int) 16000, " + "encoding-name = (string) \"SIREN\", " + "dct-length = (int) 320") + ); + +static gboolean gst_rtpsirenpay_setcaps (GstBaseRTPPayload * payload, + GstCaps * caps); + +GST_BOILERPLATE (GstRTPSirenPay, gst_rtpsirenpay, GstBaseRTPAudioPayload, + GST_TYPE_BASE_RTP_AUDIO_PAYLOAD); + +static void +gst_rtpsirenpay_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_rtpsirenpay_sink_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtpsirenpay_src_template)); + gst_element_class_set_details (element_class, &gst_rtpsirenpay_details); +} + +static void +gst_rtpsirenpay_class_init (GstRTPSirenPayClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseRTPPayloadClass *gstbasertppayload_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; + + parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD); + + gstbasertppayload_class->set_caps = gst_rtpsirenpay_setcaps; + + GST_DEBUG_CATEGORY_INIT (rtpsirenpay_debug, "rtpsirenpay", 0, + "siren audio RTP payloader"); +} + +static void +gst_rtpsirenpay_init (GstRTPSirenPay * rtpsirenpay, GstRTPSirenPayClass * klass) +{ + GstBaseRTPPayload *basertppayload; + GstBaseRTPAudioPayload *basertpaudiopayload; + + basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay); + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay); + + /* we don't set the payload type, it should be set by the application using + * the pt property or the default 96 will be used */ + basertppayload->clock_rate = 16000; + + /* tell basertpaudiopayload that this is a frame based codec */ + gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload); +} + +static gboolean +gst_rtpsirenpay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps) +{ + GstRTPSirenPay *rtpsirenpay; + GstBaseRTPAudioPayload *basertpaudiopayload; + gboolean ret; + gint dct_length; + GstStructure *structure; + const char *payload_name; + + rtpsirenpay = GST_RTP_SIREN_PAY (basertppayload); + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload); + + structure = gst_caps_get_structure (caps, 0); + + gst_structure_get_int (structure, "dct-length", &dct_length); + if (dct_length != 320) + goto wrong_dct; + + payload_name = gst_structure_get_name (structure); + if (g_strcasecmp ("audio/x-siren", payload_name)) + goto wrong_caps; + + gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN", 16000); + /* set options for this frame based audio codec */ + gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40); + + ret = gst_basertppayload_set_outcaps (basertppayload, NULL); + + return TRUE; + + /* ERRORS */ +wrong_dct: + { + GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d", dct_length); + return FALSE; + } +wrong_caps: + { + GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s", + payload_name); + return FALSE; + } +} + +gboolean +gst_rtp_siren_pay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpsirenpay", + GST_RANK_NONE, GST_TYPE_RTP_SIREN_PAY); +} |