From e4b8c514ccdb072881a7378b46d4325a368a88d6 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Tue, 10 Feb 2009 17:20:57 +0000 Subject: [PATCH] Move siren rtp pay/depay from gst-plugins-farsight --- gst/rtp/Makefile.am | 6 +- gst/rtp/gstrtp.c | 8 ++ gst/rtp/gstrtpsirendepay.c | 133 ++++++++++++++++++++++++++++++ gst/rtp/gstrtpsirendepay.h | 57 +++++++++++++ gst/rtp/gstrtpsirenpay.c | 164 +++++++++++++++++++++++++++++++++++++ gst/rtp/gstrtpsirenpay.h | 55 +++++++++++++ 6 files changed, 422 insertions(+), 1 deletion(-) create mode 100644 gst/rtp/gstrtpsirendepay.c create mode 100644 gst/rtp/gstrtpsirendepay.h create mode 100644 gst/rtp/gstrtpsirenpay.c create mode 100644 gst/rtp/gstrtpsirenpay.h diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am index 534f6108b7..679f011ab7 100644 --- a/gst/rtp/Makefile.am +++ b/gst/rtp/Makefile.am @@ -46,6 +46,8 @@ libgstrtp_la_SOURCES = \ gstrtpmp4gpay.c \ gstrtpmp4adepay.c \ gstrtpmp4apay.c \ + gstrtpsirenpay.c \ + gstrtpsirendepay.c \ gstrtpspeexdepay.c \ gstrtpspeexpay.c \ gstrtpsv3vdepay.c \ @@ -55,7 +57,7 @@ libgstrtp_la_SOURCES = \ gstrtpvorbispay.c \ gstrtpvrawdepay.c \ gstrtpvrawpay.c - + if HAVE_WINSOCK2_H WINSOCK2_LIBS = -lws2_32 @@ -116,6 +118,8 @@ noinst_HEADERS = \ gstrtpmp4apay.h \ gstrtpdepay.h \ gstasteriskh263.h \ + gstrtpsirenpay.h \ + gstrtpsirendepay.h \ gstrtpspeexdepay.h \ gstrtpspeexpay.h \ gstrtpsv3vdepay.h \ diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c index 22022c6029..fe81cc4ecb 100644 --- a/gst/rtp/gstrtp.c +++ b/gst/rtp/gstrtp.c @@ -63,6 +63,8 @@ #include "gstrtpmp4apay.h" #include "gstrtpmp4gdepay.h" #include "gstrtpmp4gpay.h" +#include "gstrtpsirenpay.h" +#include "gstrtpsirendepay.h" #include "gstrtpspeexpay.h" #include "gstrtpspeexdepay.h" #include "gstrtpsv3vdepay.h" @@ -202,6 +204,12 @@ plugin_init (GstPlugin * plugin) if (!gst_rtp_mp4g_pay_plugin_init (plugin)) return FALSE; + if (!gst_rtp_siren_pay_plugin_init (plugin)) + return FALSE; + + if (!gst_rtp_siren_depay_plugin_init (plugin)) + return FALSE; + if (!gst_rtp_speex_pay_plugin_init (plugin)) return FALSE; diff --git a/gst/rtp/gstrtpsirendepay.c b/gst/rtp/gstrtpsirendepay.c new file mode 100644 index 0000000000..4451814951 --- /dev/null +++ b/gst/rtp/gstrtpsirendepay.c @@ -0,0 +1,133 @@ +/* + * Siren Depayloader Gst Element + * + * @author: Youness Alaoui + * + * 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 +#include +#include +#include "gstrtpsirendepay.h" + +/* elementfactory information */ +static const GstElementDetails gst_rtp_siren_depay_details = +GST_ELEMENT_DETAILS ("RTP Siren packet depayloader", + "Codec/Depayloader/Network", + "Extracts Siren audio from RTP packets", + "Philippe Kalaf "); + + +static GstStaticPadTemplate gst_rtp_siren_depay_sink_template = +GST_STATIC_PAD_TEMPLATE ("sink", + 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) 16000, " + "encoding-name = (string) \"SIREN\", " "dct-length = (int) 320") + ); + +static GstStaticPadTemplate gst_rtp_siren_depay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320") + ); + +static GstBuffer *gst_rtp_siren_depay_process (GstBaseRTPDepayload * depayload, + GstBuffer * buf); +static gboolean gst_rtp_siren_depay_setcaps (GstBaseRTPDepayload * depayload, + GstCaps * caps); + +GST_BOILERPLATE (GstRTPSirenDepay, gst_rtp_siren_depay, GstBaseRTPDepayload, + GST_TYPE_BASE_RTP_DEPAYLOAD); + +static void +gst_rtp_siren_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_siren_depay_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_siren_depay_sink_template)); + gst_element_class_set_details (element_class, &gst_rtp_siren_depay_details); +} + +static void +gst_rtp_siren_depay_class_init (GstRTPSirenDepayClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseRTPDepayloadClass *gstbasertpdepayload_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; + + gstbasertpdepayload_class->process = gst_rtp_siren_depay_process; + gstbasertpdepayload_class->set_caps = gst_rtp_siren_depay_setcaps; +} + +static void +gst_rtp_siren_depay_init (GstRTPSirenDepay * rtpsirendepay, + GstRTPSirenDepayClass * klass) +{ + +} + +static gboolean +gst_rtp_siren_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) +{ + GstCaps *srccaps; + gboolean ret; + + srccaps = gst_caps_new_simple ("audio/x-siren", + "dct-length", G_TYPE_INT, 320, NULL); + ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps); + + GST_DEBUG ("set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret); + gst_caps_unref (srccaps); + + /* always fixed clock rate of 16000 */ + depayload->clock_rate = 16000; + + return ret; +} + +static GstBuffer * +gst_rtp_siren_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) +{ + GstBuffer *outbuf; + + outbuf = gst_rtp_buffer_get_payload_buffer (buf); + + return outbuf; +} + +gboolean +gst_rtp_siren_depay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpsirendepay", + GST_RANK_MARGINAL, GST_TYPE_RTP_SIREN_DEPAY); +} diff --git a/gst/rtp/gstrtpsirendepay.h b/gst/rtp/gstrtpsirendepay.h new file mode 100644 index 0000000000..78de4a7d66 --- /dev/null +++ b/gst/rtp/gstrtpsirendepay.h @@ -0,0 +1,57 @@ +/* + * Siren Depayloader Gst Element + * + * @author: Youness Alaoui + * + * 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. + */ + +#ifndef __GST_RTP_SIREN_DEPAY_H__ +#define __GST_RTP_SIREN_DEPAY_H__ + +#include +#include + +G_BEGIN_DECLS typedef struct _GstRTPSirenDepay GstRTPSirenDepay; +typedef struct _GstRTPSirenDepayClass GstRTPSirenDepayClass; + +#define GST_TYPE_RTP_SIREN_DEPAY \ + (gst_rtp_siren_depay_get_type()) +#define GST_RTP_SIREN_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SIREN_DEPAY,GstRTPSirenDepay)) +#define GST_RTP_SIREN_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SIREN_DEPAY,GstRTPSirenDepayClass)) +#define GST_IS_RTP_SIREN_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SIREN_DEPAY)) +#define GST_IS_RTP_SIREN_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SIREN_DEPAY)) + + +struct _GstRTPSirenDepay +{ + GstBaseRTPDepayload depayload; + +}; + +struct _GstRTPSirenDepayClass +{ + GstBaseRTPDepayloadClass parent_class; +}; + +gboolean gst_rtp_siren_depay_plugin_init (GstPlugin * plugin); + +G_END_DECLS +#endif /* __GST_RTP_SIREN_DEPAY_H__ */ diff --git a/gst/rtp/gstrtpsirenpay.c b/gst/rtp/gstrtpsirenpay.c new file mode 100644 index 0000000000..5dd34d2c64 --- /dev/null +++ b/gst/rtp/gstrtpsirenpay.c @@ -0,0 +1,164 @@ +/* + * Siren Payloader Gst Element + * + * @author: Youness Alaoui + * + * 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 + +/* elementfactory information */ +static GstElementDetails gst_rtpsirenpay_details = { + "RTP Payloader for Siren Audio", + "Codec/Payloader/Network", + "Packetize Siren audio streams into RTP packets", + "Youness Alaoui " +}; + +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); +} diff --git a/gst/rtp/gstrtpsirenpay.h b/gst/rtp/gstrtpsirenpay.h new file mode 100644 index 0000000000..71eb83937f --- /dev/null +++ b/gst/rtp/gstrtpsirenpay.h @@ -0,0 +1,55 @@ +/* + * Siren Payloader Gst Element + * + * @author: Youness Alaoui + * + * 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. + */ + +#ifndef __GST_RTP_SIREN_PAY_H__ +#define __GST_RTP_SIREN_PAY_H__ + +#include +#include + +G_BEGIN_DECLS +#define GST_TYPE_RTP_SIREN_PAY \ + (gst_rtpsirenpay_get_type()) +#define GST_RTP_SIREN_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SIREN_PAY,GstRTPSirenPay)) +#define GST_RTP_SIREN_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SIREN_PAY,GstRTPSirenPayClass)) +#define GST_IS_RTP_SIREN_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SIREN_PAY)) +#define GST_IS_RTP_SIREN_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SIREN_PAY)) +typedef struct _GstRTPSirenPay GstRTPSirenPay; +typedef struct _GstRTPSirenPayClass GstRTPSirenPayClass; + +struct _GstRTPSirenPay +{ + GstBaseRTPAudioPayload audiopayload; +}; + +struct _GstRTPSirenPayClass +{ + GstBaseRTPAudioPayloadClass parent_class; +}; + +gboolean gst_rtp_siren_pay_plugin_init (GstPlugin * plugin); + +G_END_DECLS +#endif /* __GST_RTP_SIREN_PAY_H__ */