From 333468b5d448062b40c428e5c3b95dcb8159807c Mon Sep 17 00:00:00 2001 From: Zeeshan Ali Date: Sun, 13 Jul 2003 00:20:44 +0000 Subject: [PATCH] GSM -> RTP and viceversa Original commit message from CVS: GSM -> RTP and viceversa --- gst/rtp/Makefile.am | 4 +- gst/rtp/gstrtp.c | 4 + gst/rtp/gstrtpL16depay.c | 6 +- gst/rtp/gstrtpL16enc.c | 6 +- gst/rtp/gstrtpL16parse.c | 6 +- gst/rtp/gstrtpL16pay.c | 6 +- gst/rtp/gstrtpgsmdepay.c | 295 +++++++++++++++++++++++++++++++++++++ gst/rtp/gstrtpgsmdepay.h | 70 +++++++++ gst/rtp/gstrtpgsmenc.c | 310 +++++++++++++++++++++++++++++++++++++++ gst/rtp/gstrtpgsmenc.h | 79 ++++++++++ gst/rtp/gstrtpgsmparse.c | 295 +++++++++++++++++++++++++++++++++++++ gst/rtp/gstrtpgsmparse.h | 70 +++++++++ gst/rtp/gstrtpgsmpay.c | 310 +++++++++++++++++++++++++++++++++++++++ gst/rtp/gstrtpgsmpay.h | 79 ++++++++++ gst/rtp/rtp-packet.c | 10 +- gst/rtp/rtp-packet.h | 14 +- 16 files changed, 1536 insertions(+), 28 deletions(-) create mode 100644 gst/rtp/gstrtpgsmdepay.c create mode 100644 gst/rtp/gstrtpgsmdepay.h create mode 100644 gst/rtp/gstrtpgsmenc.c create mode 100644 gst/rtp/gstrtpgsmenc.h create mode 100644 gst/rtp/gstrtpgsmparse.c create mode 100644 gst/rtp/gstrtpgsmparse.h create mode 100644 gst/rtp/gstrtpgsmpay.c create mode 100644 gst/rtp/gstrtpgsmpay.h diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am index 573d007718..83bcbd316a 100644 --- a/gst/rtp/Makefile.am +++ b/gst/rtp/Makefile.am @@ -2,9 +2,9 @@ plugindir = $(libdir)/gstreamer-@GST_MAJORMINOR@ plugin_LTLIBRARIES = libgstrtp.la -libgstrtp_la_SOURCES = gstrtp.c gstrtpL16enc.c gstrtpL16parse.c rtp-packet.c +libgstrtp_la_SOURCES = gstrtp.c gstrtpL16enc.c gstrtpL16parse.c gstrtpgsmenc.c gstrtpgsmparse.c rtp-packet.c libgstrtp_la_CFLAGS = $(GST_CFLAGS) libgstrtp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -noinst_HEADERS = gstrtpL16enc.h gstrtpL16parse.h gstrtp-common.h rtp-packet.h +noinst_HEADERS = gstrtpL16enc.h gstrtpL16parse.h gstrtpgsmenc.h gstrtpgsmparse.h gstrtp-common.h rtp-packet.h diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c index 063dcc7711..2cf33157a8 100644 --- a/gst/rtp/gstrtp.c +++ b/gst/rtp/gstrtp.c @@ -19,12 +19,16 @@ #include "gstrtpL16enc.h" #include "gstrtpL16parse.h" +#include "gstrtpgsmenc.h" +#include "gstrtpgsmparse.h" static gboolean plugin_init (GModule *module, GstPlugin *plugin) { gst_rtpL16enc_plugin_init (module, plugin); gst_rtpL16parse_plugin_init (module, plugin); + gst_rtpgsmenc_plugin_init (module, plugin); + gst_rtpgsmparse_plugin_init (module, plugin); return TRUE; } diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c index fc3355c07c..832bb23d66 100644 --- a/gst/rtp/gstrtpL16depay.c +++ b/gst/rtp/gstrtpL16depay.c @@ -146,11 +146,11 @@ gst_rtpL16parse_init (GstRtpL16Parse * rtpL16parse) void gst_rtpL16parse_ntohs (GstBuffer *buf) { - guint16 *i, *len; + gint16 *i, *len; /* FIXME: is this code correct or even sane at all? */ - i = (guint16 *) GST_BUFFER_DATA(buf); - len = i + GST_BUFFER_SIZE (buf) / sizeof (guint16 *); + i = (gint16 *) GST_BUFFER_DATA(buf); + len = i + GST_BUFFER_SIZE (buf) / sizeof (gint16 *); for (; i Erik Walthinsen + * + * 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 + */ + +#include +#include "gstrtpgsmparse.h" +#include "gstrtp-common.h" + +/* elementfactory information */ +static GstElementDetails gst_rtp_L16parse_details = { + "RTP packet parser", + "RtpGSMParse", + "GPL", + "Extracts GSM audio from RTP packets", + VERSION, + "Zeeshan Ali ", + "(C) 2003", +}; + +/* RtpGSMParse signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + ARG_0, + ARG_FREQUENCY +}; + +GST_PAD_TEMPLATE_FACTORY (src_factory, + "src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "gsm_gsm", + "audio/x-gsm", + "rate", GST_PROPS_INT_RANGE (1000, 48000)) +) + +GST_PAD_TEMPLATE_FACTORY (sink_factory, + "sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "rtp", + "application/x-rtp", + NULL) +); + +static void gst_rtpgsmparse_class_init (GstRtpGSMParseClass * klass); +static void gst_rtpgsmparse_init (GstRtpGSMParse * rtpgsmparse); + +static void gst_rtpgsmparse_chain (GstPad * pad, GstBuffer * buf); + +static void gst_rtpgsmparse_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_rtpgsmparse_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); +static GstElementStateReturn gst_rtpgsmparse_change_state (GstElement * element); + +static GstElementClass *parent_class = NULL; + +static GType gst_rtpgsmparse_get_type (void) +{ + static GType rtpgsmparse_type = 0; + + if (!rtpgsmparse_type) { + static const GTypeInfo rtpgsmparse_info = { + sizeof (GstRtpGSMParseClass), + NULL, + NULL, + (GClassInitFunc) gst_rtpgsmparse_class_init, + NULL, + NULL, + sizeof (GstRtpGSMParse), + 0, + (GInstanceInitFunc) gst_rtpgsmparse_init, + }; + + rtpgsmparse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRtpGSMParse", &rtpgsmparse_info, 0); + } + return rtpgsmparse_type; +} + +static void +gst_rtpgsmparse_class_init (GstRtpGSMParseClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + + parent_class = g_type_class_ref (GST_TYPE_ELEMENT); + + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FREQUENCY, + g_param_spec_int ("frequency", "frequency", "frequency", + G_MININT, G_MAXINT, 8000, G_PARAM_READWRITE)); + + gobject_class->set_property = gst_rtpgsmparse_set_property; + gobject_class->get_property = gst_rtpgsmparse_get_property; + + gstelement_class->change_state = gst_rtpgsmparse_change_state; +} + +static void +gst_rtpgsmparse_init (GstRtpGSMParse * rtpgsmparse) +{ + rtpgsmparse->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_factory), "src"); + rtpgsmparse->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_factory), "sink"); + gst_element_add_pad (GST_ELEMENT (rtpgsmparse), rtpgsmparse->srcpad); + gst_element_add_pad (GST_ELEMENT (rtpgsmparse), rtpgsmparse->sinkpad); + gst_pad_set_chain_function (rtpgsmparse->sinkpad, gst_rtpgsmparse_chain); + + rtpgsmparse->frequency = 8000; +} + +void +gst_rtpgsmparse_ntohs (GstBuffer *buf) +{ + gint16 *i, *len; + + /* FIXME: is this code correct or even sane at all? */ + i = (gint16 *) GST_BUFFER_DATA(buf); + len = i + GST_BUFFER_SIZE (buf) / sizeof (gint16 *); + + for (; ifrequency)); + + gst_pad_try_set_caps (rtpgsmparse->srcpad, caps); +} + +static void +gst_rtpgsmparse_chain (GstPad * pad, GstBuffer * buf) +{ + GstRtpGSMParse *rtpgsmparse; + GstBuffer *outbuf; + Rtp_Packet packet; + rtp_payload_t pt; + + g_return_if_fail (pad != NULL); + g_return_if_fail (GST_IS_PAD (pad)); + g_return_if_fail (buf != NULL); + + rtpgsmparse = GST_RTP_GSM_PARSE (GST_OBJECT_PARENT (pad)); + + g_return_if_fail (rtpgsmparse != NULL); + g_return_if_fail (GST_IS_RTP_GSM_PARSE (rtpgsmparse)); + + if (GST_IS_EVENT (buf)) { + GstEvent *event = GST_EVENT (buf); + gst_pad_event_default (pad, event); + + return; + } + + if (GST_PAD_CAPS (rtpgsmparse->srcpad) == NULL) { + gst_rtpgsm_caps_nego (rtpgsmparse); + } + + packet = rtp_packet_new_copy_data (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); + + pt = rtp_packet_get_payload_type (packet); + + if (pt != PAYLOAD_GSM) { + g_warning ("Unexpected paload type %u\n", pt); + rtp_packet_free (packet); + gst_buffer_unref (buf); + return; + } + + outbuf = gst_buffer_new (); + GST_BUFFER_SIZE (outbuf) = rtp_packet_get_payload_len (packet); + GST_BUFFER_DATA (outbuf) = g_malloc (GST_BUFFER_SIZE (outbuf)); + GST_BUFFER_TIMESTAMP (outbuf) = g_ntohl (rtp_packet_get_timestamp (packet)) * GST_SECOND; + + memcpy (GST_BUFFER_DATA (outbuf), rtp_packet_get_payload (packet), GST_BUFFER_SIZE (outbuf)); + + GST_DEBUG (0,"gst_rtpgsmparse_chain: pushing buffer of size %d", GST_BUFFER_SIZE(outbuf)); + +/* FIXME: According to RFC 1890, this is required, right? */ +#if G_BYTE_ORDER == G_LITTLE_ENDIAN + gst_rtpgsmparse_ntohs (outbuf); +#endif + + gst_pad_push (rtpgsmparse->srcpad, outbuf); + + rtp_packet_free (packet); + gst_buffer_unref (buf); +} + +static void +gst_rtpgsmparse_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) +{ + GstRtpGSMParse *rtpgsmparse; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_RTP_GSM_PARSE (object)); + rtpgsmparse = GST_RTP_GSM_PARSE (object); + + switch (prop_id) { + case ARG_FREQUENCY: + rtpgsmparse->frequency = g_value_get_int (value); + break; + default: + break; + } +} + +static void +gst_rtpgsmparse_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) +{ + GstRtpGSMParse *rtpgsmparse; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_RTP_GSM_PARSE (object)); + rtpgsmparse = GST_RTP_GSM_PARSE (object); + + switch (prop_id) { + case ARG_FREQUENCY: + g_value_set_int (value, rtpgsmparse->frequency); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstElementStateReturn +gst_rtpgsmparse_change_state (GstElement * element) +{ + GstRtpGSMParse *rtpgsmparse; + + g_return_val_if_fail (GST_IS_RTP_GSM_PARSE (element), GST_STATE_FAILURE); + + rtpgsmparse = GST_RTP_GSM_PARSE (element); + + GST_DEBUG (0, "state pending %d\n", GST_STATE_PENDING (element)); + + switch (GST_STATE_TRANSITION (element)) { + case GST_STATE_NULL_TO_READY: + break; + case GST_STATE_READY_TO_NULL: + break; + default: + break; + } + + /* if we haven't failed already, give the parent class a chance to ;-) */ + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + + return GST_STATE_SUCCESS; +} + +gboolean +gst_rtpgsmparse_plugin_init (GModule * module, GstPlugin * plugin) +{ + GstElementFactory *rtpgsmparse; + + rtpgsmparse = gst_element_factory_new ("rtpgsmparse", GST_TYPE_RTP_GSM_PARSE, &gst_rtp_L16parse_details); + g_return_val_if_fail (rtpgsmparse != NULL, FALSE); + + gst_element_factory_add_pad_template (rtpgsmparse, GST_PAD_TEMPLATE_GET (src_factory)); + gst_element_factory_add_pad_template (rtpgsmparse, GST_PAD_TEMPLATE_GET (sink_factory)); + + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (rtpgsmparse)); + + return TRUE; +} diff --git a/gst/rtp/gstrtpgsmdepay.h b/gst/rtp/gstrtpgsmdepay.h new file mode 100644 index 0000000000..9d61fb7253 --- /dev/null +++ b/gst/rtp/gstrtpgsmdepay.h @@ -0,0 +1,70 @@ +/* Gnome-Streamer + * Copyright (C) <1999> Erik Walthinsen + * + * 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_GSM_PARSE_H__ +#define __GST_RTP_GSM_PARSE_H__ + +#include +#include "rtp-packet.h" +#include "gstrtp-common.h" + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + +/* Definition of structure storing data for this element. */ +typedef struct _GstRtpGSMParse GstRtpGSMParse; +struct _GstRtpGSMParse +{ + GstElement element; + + GstPad *sinkpad; + GstPad *srcpad; + + guint frequency; +}; + +/* Standard definition defining a class for this element. */ +typedef struct _GstRtpGSMParseClass GstRtpGSMParseClass; +struct _GstRtpGSMParseClass +{ + GstElementClass parent_class; +}; + +/* Standard macros for defining types for this element. */ +#define GST_TYPE_RTP_GSM_PARSE \ + (gst_rtpgsmparse_get_type()) +#define GST_RTP_GSM_PARSE(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_GSM_PARSE,GstRtpGSMParse)) +#define GST_RTP_GSM_PARSE_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_GSM_PARSE,GstRtpGSMParse)) +#define GST_IS_RTP_GSM_PARSE(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_GSM_PARSE)) +#define GST_IS_RTP_GSM_PARSE_CLASS(obj) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_GSM_PARSE)) + +gboolean gst_rtpgsmparse_plugin_init (GModule * module, GstPlugin * plugin); + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + + +#endif /* __GST_RTP_GSM_PARSE_H__ */ diff --git a/gst/rtp/gstrtpgsmenc.c b/gst/rtp/gstrtpgsmenc.c new file mode 100644 index 0000000000..2fff85c6f8 --- /dev/null +++ b/gst/rtp/gstrtpgsmenc.c @@ -0,0 +1,310 @@ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen + * + * 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. + */ + +#include +#include "gstrtpgsmenc.h" + +/* elementfactory information */ +static GstElementDetails gst_rtpgsmenc_details = { + "RTP GSM Audio Encoder", + "RtpGSMEnc", + "LGPL", + "Encodes GSM audio into an RTP packet", + VERSION, + "Zeeshan Ali ", + "(C) 2003", +}; + +/* RtpGSMEnc signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + /* FILL ME */ + ARG_0, +}; + +GST_PAD_TEMPLATE_FACTORY (sink_factory, + "sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "gsm_gsm", + "audio/x-gsm", + "rate", GST_PROPS_INT_RANGE (1000, 48000) + ) +); + +GST_PAD_TEMPLATE_FACTORY (src_factory, + "src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "rtp", + "application/x-rtp", + NULL) +); + +static void gst_rtpgsmenc_class_init (GstRtpGSMEncClass * klass); +static void gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc); +static void gst_rtpgsmenc_chain (GstPad * pad, GstBuffer * buf); +static void gst_rtpgsmenc_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_rtpgsmenc_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); +static GstPadLinkReturn gst_rtpgsmenc_sinkconnect (GstPad * pad, GstCaps * caps); +static GstElementStateReturn gst_rtpgsmenc_change_state (GstElement * element); + +static GstElementClass *parent_class = NULL; + +static GType gst_rtpgsmenc_get_type (void) +{ + static GType rtpgsmenc_type = 0; + + if (!rtpgsmenc_type) { + static const GTypeInfo rtpgsmenc_info = { + sizeof (GstRtpGSMEncClass), + NULL, + NULL, + (GClassInitFunc) gst_rtpgsmenc_class_init, + NULL, + NULL, + sizeof (GstRtpGSMEnc), + 0, + (GInstanceInitFunc) gst_rtpgsmenc_init, + }; + + rtpgsmenc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRtpGSMEnc", &rtpgsmenc_info, 0); + } + return rtpgsmenc_type; +} + +static void +gst_rtpgsmenc_class_init (GstRtpGSMEncClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + + parent_class = g_type_class_ref (GST_TYPE_ELEMENT); + + gobject_class->set_property = gst_rtpgsmenc_set_property; + gobject_class->get_property = gst_rtpgsmenc_get_property; + + gstelement_class->change_state = gst_rtpgsmenc_change_state; +} + +static void +gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc) +{ + rtpgsmenc->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_factory), "sink"); + rtpgsmenc->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_factory), "src"); + gst_element_add_pad (GST_ELEMENT (rtpgsmenc), rtpgsmenc->sinkpad); + gst_element_add_pad (GST_ELEMENT (rtpgsmenc), rtpgsmenc->srcpad); + gst_pad_set_chain_function (rtpgsmenc->sinkpad, gst_rtpgsmenc_chain); + gst_pad_set_link_function (rtpgsmenc->sinkpad, gst_rtpgsmenc_sinkconnect); + + rtpgsmenc->frequency = 8000; + + rtpgsmenc->next_time = 0; + rtpgsmenc->time_interval = 0; + + rtpgsmenc->seq = 0; + rtpgsmenc->ssrc = random (); +} + +static GstPadLinkReturn +gst_rtpgsmenc_sinkconnect (GstPad * pad, GstCaps * caps) +{ + GstRtpGSMEnc *rtpgsmenc; + + rtpgsmenc = GST_RTP_GSM_ENC (gst_pad_get_parent (pad)); + + gst_caps_get_int (caps, "rate", &rtpgsmenc->frequency); + + /* Pre-calculate what we can */ + rtpgsmenc->time_interval = GST_SECOND / (2 * rtpgsmenc->frequency); + + return GST_PAD_LINK_OK; +} + + +void +gst_rtpgsmenc_htons (GstBuffer *buf) +{ + gint16 *i, *len; + + /* FIXME: is this code correct or even sane at all? */ + i = (gint16 *) GST_BUFFER_DATA(buf); + len = i + GST_BUFFER_SIZE (buf) / sizeof (gint16 *); + + for (; inext_time = 0; + gst_pad_event_default (pad, event); + return; + default: + gst_pad_event_default (pad, event); + return; + } + } + + /* We only need the header */ + packet = rtp_packet_new_allocate (0, 0, 0); + + rtp_packet_set_csrc_count (packet, 0); + rtp_packet_set_extension (packet, 0); + rtp_packet_set_padding (packet, 0); + rtp_packet_set_version (packet, RTP_VERSION); + rtp_packet_set_marker (packet, 0); + rtp_packet_set_ssrc (packet, g_htonl (rtpgsmenc->ssrc)); + rtp_packet_set_seq (packet, g_htons (rtpgsmenc->seq)); + rtp_packet_set_timestamp (packet, g_htonl ((guint32) rtpgsmenc->next_time / GST_SECOND)); + rtp_packet_set_payload_type (packet, (guint8) PAYLOAD_GSM); + + /* FIXME: According to RFC 1890, this is required, right? */ +#if G_BYTE_ORDER == G_LITTLE_ENDIAN + gst_rtpgsmenc_htons (buf); +#endif + + outbuf = gst_buffer_new (); + GST_BUFFER_SIZE (outbuf) = rtp_packet_get_packet_len (packet) + GST_BUFFER_SIZE (buf); + GST_BUFFER_DATA (outbuf) = g_malloc (GST_BUFFER_SIZE (outbuf)); + GST_BUFFER_TIMESTAMP (outbuf) = rtpgsmenc->next_time; + + memcpy (GST_BUFFER_DATA (outbuf), packet->data, rtp_packet_get_packet_len (packet)); + memcpy (GST_BUFFER_DATA (outbuf) + rtp_packet_get_packet_len(packet), GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); + + GST_DEBUG (0,"gst_rtpgsmenc_chain: pushing buffer of size %d", GST_BUFFER_SIZE(outbuf)); + gst_pad_push (rtpgsmenc->srcpad, outbuf); + + ++rtpgsmenc->seq; + rtpgsmenc->next_time += rtpgsmenc->time_interval * GST_BUFFER_SIZE (buf); + + rtp_packet_free (packet); + gst_buffer_unref (buf); +} + +static void +gst_rtpgsmenc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) +{ + GstRtpGSMEnc *rtpgsmenc; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_RTP_GSM_ENC (object)); + rtpgsmenc = GST_RTP_GSM_ENC (object); + + switch (prop_id) { + default: + break; + } +} + +static void +gst_rtpgsmenc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) +{ + GstRtpGSMEnc *rtpgsmenc; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_RTP_GSM_ENC (object)); + rtpgsmenc = GST_RTP_GSM_ENC (object); + + switch (prop_id) { + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstElementStateReturn +gst_rtpgsmenc_change_state (GstElement * element) +{ + GstRtpGSMEnc *rtpgsmenc; + + g_return_val_if_fail (GST_IS_RTP_GSM_ENC (element), GST_STATE_FAILURE); + + rtpgsmenc = GST_RTP_GSM_ENC (element); + + GST_DEBUG (0, "state pending %d\n", GST_STATE_PENDING (element)); + + /* if going down into NULL state, close the file if it's open */ + switch (GST_STATE_TRANSITION (element)) { + case GST_STATE_NULL_TO_READY: + break; + + case GST_STATE_READY_TO_NULL: + break; + + default: + break; + } + + /* if we haven't failed already, give the parent class a chance to ;-) */ + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + + return GST_STATE_SUCCESS; +} + +gboolean +gst_rtpgsmenc_plugin_init (GModule * module, GstPlugin * plugin) +{ + GstElementFactory *rtpgsmenc; + + rtpgsmenc = gst_element_factory_new ("rtpgsmenc", GST_TYPE_RTP_GSM_ENC, &gst_rtpgsmenc_details); + g_return_val_if_fail (rtpgsmenc != NULL, FALSE); + + gst_element_factory_add_pad_template (rtpgsmenc, GST_PAD_TEMPLATE_GET (sink_factory)); + gst_element_factory_add_pad_template (rtpgsmenc, GST_PAD_TEMPLATE_GET (src_factory)); + + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (rtpgsmenc)); + + return TRUE; +} diff --git a/gst/rtp/gstrtpgsmenc.h b/gst/rtp/gstrtpgsmenc.h new file mode 100644 index 0000000000..76a0ec0e6d --- /dev/null +++ b/gst/rtp/gstrtpgsmenc.h @@ -0,0 +1,79 @@ +/* Gnome-Streamer + * Copyright (C) <1999> Erik Walthinsen + * + * 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_GSM_ENC_H__ +#define __GST_RTP_GSM_ENC_H__ + +#include +#include "rtp-packet.h" +#include "gstrtp-common.h" + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + +/* Definition of structure storing data for this element. */ +typedef struct _GstRtpGSMEnc GstRtpGSMEnc; +struct _GstRtpGSMEnc +{ + GstElement element; + + GstPad *sinkpad; + GstPad *srcpad; + + guint frequency; + + /* the timestamp of the next frame */ + guint64 next_time; + /* the interval between frames */ + guint64 time_interval; + + guint32 ssrc; + guint16 seq; +}; + +/* Standard definition defining a class for this element. */ +typedef struct _GstRtpGSMEncClass GstRtpGSMEncClass; +struct _GstRtpGSMEncClass +{ + GstElementClass parent_class; +}; + +/* Standard macros for defining types for this element. */ +#define GST_TYPE_RTP_GSM_ENC \ + (gst_rtpgsmenc_get_type()) +#define GST_RTP_GSM_ENC(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_GSM_ENC,GstRtpGSMEnc)) +#define GST_RTP_GSM_ENC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_GSM_ENC,GstRtpGSMEnc)) +#define GST_IS_RTP_GSM_ENC(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_GSM_ENC)) +#define GST_IS_RTP_GSM_ENC_CLASS(obj) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_GSM_ENC)) + +gboolean gst_rtpgsmenc_plugin_init (GModule * module, GstPlugin * plugin); + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + + +#endif /* __GST_RTP_GSM_ENC_H__ */ diff --git a/gst/rtp/gstrtpgsmparse.c b/gst/rtp/gstrtpgsmparse.c new file mode 100644 index 0000000000..814cc6804c --- /dev/null +++ b/gst/rtp/gstrtpgsmparse.c @@ -0,0 +1,295 @@ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen + * + * 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 + */ + +#include +#include "gstrtpgsmparse.h" +#include "gstrtp-common.h" + +/* elementfactory information */ +static GstElementDetails gst_rtp_L16parse_details = { + "RTP packet parser", + "RtpGSMParse", + "GPL", + "Extracts GSM audio from RTP packets", + VERSION, + "Zeeshan Ali ", + "(C) 2003", +}; + +/* RtpGSMParse signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + ARG_0, + ARG_FREQUENCY +}; + +GST_PAD_TEMPLATE_FACTORY (src_factory, + "src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "gsm_gsm", + "audio/x-gsm", + "rate", GST_PROPS_INT_RANGE (1000, 48000)) +) + +GST_PAD_TEMPLATE_FACTORY (sink_factory, + "sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "rtp", + "application/x-rtp", + NULL) +); + +static void gst_rtpgsmparse_class_init (GstRtpGSMParseClass * klass); +static void gst_rtpgsmparse_init (GstRtpGSMParse * rtpgsmparse); + +static void gst_rtpgsmparse_chain (GstPad * pad, GstBuffer * buf); + +static void gst_rtpgsmparse_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_rtpgsmparse_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); +static GstElementStateReturn gst_rtpgsmparse_change_state (GstElement * element); + +static GstElementClass *parent_class = NULL; + +static GType gst_rtpgsmparse_get_type (void) +{ + static GType rtpgsmparse_type = 0; + + if (!rtpgsmparse_type) { + static const GTypeInfo rtpgsmparse_info = { + sizeof (GstRtpGSMParseClass), + NULL, + NULL, + (GClassInitFunc) gst_rtpgsmparse_class_init, + NULL, + NULL, + sizeof (GstRtpGSMParse), + 0, + (GInstanceInitFunc) gst_rtpgsmparse_init, + }; + + rtpgsmparse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRtpGSMParse", &rtpgsmparse_info, 0); + } + return rtpgsmparse_type; +} + +static void +gst_rtpgsmparse_class_init (GstRtpGSMParseClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + + parent_class = g_type_class_ref (GST_TYPE_ELEMENT); + + g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FREQUENCY, + g_param_spec_int ("frequency", "frequency", "frequency", + G_MININT, G_MAXINT, 8000, G_PARAM_READWRITE)); + + gobject_class->set_property = gst_rtpgsmparse_set_property; + gobject_class->get_property = gst_rtpgsmparse_get_property; + + gstelement_class->change_state = gst_rtpgsmparse_change_state; +} + +static void +gst_rtpgsmparse_init (GstRtpGSMParse * rtpgsmparse) +{ + rtpgsmparse->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_factory), "src"); + rtpgsmparse->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_factory), "sink"); + gst_element_add_pad (GST_ELEMENT (rtpgsmparse), rtpgsmparse->srcpad); + gst_element_add_pad (GST_ELEMENT (rtpgsmparse), rtpgsmparse->sinkpad); + gst_pad_set_chain_function (rtpgsmparse->sinkpad, gst_rtpgsmparse_chain); + + rtpgsmparse->frequency = 8000; +} + +void +gst_rtpgsmparse_ntohs (GstBuffer *buf) +{ + gint16 *i, *len; + + /* FIXME: is this code correct or even sane at all? */ + i = (gint16 *) GST_BUFFER_DATA(buf); + len = i + GST_BUFFER_SIZE (buf) / sizeof (gint16 *); + + for (; ifrequency)); + + gst_pad_try_set_caps (rtpgsmparse->srcpad, caps); +} + +static void +gst_rtpgsmparse_chain (GstPad * pad, GstBuffer * buf) +{ + GstRtpGSMParse *rtpgsmparse; + GstBuffer *outbuf; + Rtp_Packet packet; + rtp_payload_t pt; + + g_return_if_fail (pad != NULL); + g_return_if_fail (GST_IS_PAD (pad)); + g_return_if_fail (buf != NULL); + + rtpgsmparse = GST_RTP_GSM_PARSE (GST_OBJECT_PARENT (pad)); + + g_return_if_fail (rtpgsmparse != NULL); + g_return_if_fail (GST_IS_RTP_GSM_PARSE (rtpgsmparse)); + + if (GST_IS_EVENT (buf)) { + GstEvent *event = GST_EVENT (buf); + gst_pad_event_default (pad, event); + + return; + } + + if (GST_PAD_CAPS (rtpgsmparse->srcpad) == NULL) { + gst_rtpgsm_caps_nego (rtpgsmparse); + } + + packet = rtp_packet_new_copy_data (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); + + pt = rtp_packet_get_payload_type (packet); + + if (pt != PAYLOAD_GSM) { + g_warning ("Unexpected paload type %u\n", pt); + rtp_packet_free (packet); + gst_buffer_unref (buf); + return; + } + + outbuf = gst_buffer_new (); + GST_BUFFER_SIZE (outbuf) = rtp_packet_get_payload_len (packet); + GST_BUFFER_DATA (outbuf) = g_malloc (GST_BUFFER_SIZE (outbuf)); + GST_BUFFER_TIMESTAMP (outbuf) = g_ntohl (rtp_packet_get_timestamp (packet)) * GST_SECOND; + + memcpy (GST_BUFFER_DATA (outbuf), rtp_packet_get_payload (packet), GST_BUFFER_SIZE (outbuf)); + + GST_DEBUG (0,"gst_rtpgsmparse_chain: pushing buffer of size %d", GST_BUFFER_SIZE(outbuf)); + +/* FIXME: According to RFC 1890, this is required, right? */ +#if G_BYTE_ORDER == G_LITTLE_ENDIAN + gst_rtpgsmparse_ntohs (outbuf); +#endif + + gst_pad_push (rtpgsmparse->srcpad, outbuf); + + rtp_packet_free (packet); + gst_buffer_unref (buf); +} + +static void +gst_rtpgsmparse_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) +{ + GstRtpGSMParse *rtpgsmparse; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_RTP_GSM_PARSE (object)); + rtpgsmparse = GST_RTP_GSM_PARSE (object); + + switch (prop_id) { + case ARG_FREQUENCY: + rtpgsmparse->frequency = g_value_get_int (value); + break; + default: + break; + } +} + +static void +gst_rtpgsmparse_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) +{ + GstRtpGSMParse *rtpgsmparse; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_RTP_GSM_PARSE (object)); + rtpgsmparse = GST_RTP_GSM_PARSE (object); + + switch (prop_id) { + case ARG_FREQUENCY: + g_value_set_int (value, rtpgsmparse->frequency); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstElementStateReturn +gst_rtpgsmparse_change_state (GstElement * element) +{ + GstRtpGSMParse *rtpgsmparse; + + g_return_val_if_fail (GST_IS_RTP_GSM_PARSE (element), GST_STATE_FAILURE); + + rtpgsmparse = GST_RTP_GSM_PARSE (element); + + GST_DEBUG (0, "state pending %d\n", GST_STATE_PENDING (element)); + + switch (GST_STATE_TRANSITION (element)) { + case GST_STATE_NULL_TO_READY: + break; + case GST_STATE_READY_TO_NULL: + break; + default: + break; + } + + /* if we haven't failed already, give the parent class a chance to ;-) */ + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + + return GST_STATE_SUCCESS; +} + +gboolean +gst_rtpgsmparse_plugin_init (GModule * module, GstPlugin * plugin) +{ + GstElementFactory *rtpgsmparse; + + rtpgsmparse = gst_element_factory_new ("rtpgsmparse", GST_TYPE_RTP_GSM_PARSE, &gst_rtp_L16parse_details); + g_return_val_if_fail (rtpgsmparse != NULL, FALSE); + + gst_element_factory_add_pad_template (rtpgsmparse, GST_PAD_TEMPLATE_GET (src_factory)); + gst_element_factory_add_pad_template (rtpgsmparse, GST_PAD_TEMPLATE_GET (sink_factory)); + + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (rtpgsmparse)); + + return TRUE; +} diff --git a/gst/rtp/gstrtpgsmparse.h b/gst/rtp/gstrtpgsmparse.h new file mode 100644 index 0000000000..9d61fb7253 --- /dev/null +++ b/gst/rtp/gstrtpgsmparse.h @@ -0,0 +1,70 @@ +/* Gnome-Streamer + * Copyright (C) <1999> Erik Walthinsen + * + * 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_GSM_PARSE_H__ +#define __GST_RTP_GSM_PARSE_H__ + +#include +#include "rtp-packet.h" +#include "gstrtp-common.h" + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + +/* Definition of structure storing data for this element. */ +typedef struct _GstRtpGSMParse GstRtpGSMParse; +struct _GstRtpGSMParse +{ + GstElement element; + + GstPad *sinkpad; + GstPad *srcpad; + + guint frequency; +}; + +/* Standard definition defining a class for this element. */ +typedef struct _GstRtpGSMParseClass GstRtpGSMParseClass; +struct _GstRtpGSMParseClass +{ + GstElementClass parent_class; +}; + +/* Standard macros for defining types for this element. */ +#define GST_TYPE_RTP_GSM_PARSE \ + (gst_rtpgsmparse_get_type()) +#define GST_RTP_GSM_PARSE(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_GSM_PARSE,GstRtpGSMParse)) +#define GST_RTP_GSM_PARSE_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_GSM_PARSE,GstRtpGSMParse)) +#define GST_IS_RTP_GSM_PARSE(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_GSM_PARSE)) +#define GST_IS_RTP_GSM_PARSE_CLASS(obj) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_GSM_PARSE)) + +gboolean gst_rtpgsmparse_plugin_init (GModule * module, GstPlugin * plugin); + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + + +#endif /* __GST_RTP_GSM_PARSE_H__ */ diff --git a/gst/rtp/gstrtpgsmpay.c b/gst/rtp/gstrtpgsmpay.c new file mode 100644 index 0000000000..2fff85c6f8 --- /dev/null +++ b/gst/rtp/gstrtpgsmpay.c @@ -0,0 +1,310 @@ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen + * + * 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. + */ + +#include +#include "gstrtpgsmenc.h" + +/* elementfactory information */ +static GstElementDetails gst_rtpgsmenc_details = { + "RTP GSM Audio Encoder", + "RtpGSMEnc", + "LGPL", + "Encodes GSM audio into an RTP packet", + VERSION, + "Zeeshan Ali ", + "(C) 2003", +}; + +/* RtpGSMEnc signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + /* FILL ME */ + ARG_0, +}; + +GST_PAD_TEMPLATE_FACTORY (sink_factory, + "sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "gsm_gsm", + "audio/x-gsm", + "rate", GST_PROPS_INT_RANGE (1000, 48000) + ) +); + +GST_PAD_TEMPLATE_FACTORY (src_factory, + "src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "rtp", + "application/x-rtp", + NULL) +); + +static void gst_rtpgsmenc_class_init (GstRtpGSMEncClass * klass); +static void gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc); +static void gst_rtpgsmenc_chain (GstPad * pad, GstBuffer * buf); +static void gst_rtpgsmenc_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_rtpgsmenc_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); +static GstPadLinkReturn gst_rtpgsmenc_sinkconnect (GstPad * pad, GstCaps * caps); +static GstElementStateReturn gst_rtpgsmenc_change_state (GstElement * element); + +static GstElementClass *parent_class = NULL; + +static GType gst_rtpgsmenc_get_type (void) +{ + static GType rtpgsmenc_type = 0; + + if (!rtpgsmenc_type) { + static const GTypeInfo rtpgsmenc_info = { + sizeof (GstRtpGSMEncClass), + NULL, + NULL, + (GClassInitFunc) gst_rtpgsmenc_class_init, + NULL, + NULL, + sizeof (GstRtpGSMEnc), + 0, + (GInstanceInitFunc) gst_rtpgsmenc_init, + }; + + rtpgsmenc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRtpGSMEnc", &rtpgsmenc_info, 0); + } + return rtpgsmenc_type; +} + +static void +gst_rtpgsmenc_class_init (GstRtpGSMEncClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + + parent_class = g_type_class_ref (GST_TYPE_ELEMENT); + + gobject_class->set_property = gst_rtpgsmenc_set_property; + gobject_class->get_property = gst_rtpgsmenc_get_property; + + gstelement_class->change_state = gst_rtpgsmenc_change_state; +} + +static void +gst_rtpgsmenc_init (GstRtpGSMEnc * rtpgsmenc) +{ + rtpgsmenc->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_factory), "sink"); + rtpgsmenc->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_factory), "src"); + gst_element_add_pad (GST_ELEMENT (rtpgsmenc), rtpgsmenc->sinkpad); + gst_element_add_pad (GST_ELEMENT (rtpgsmenc), rtpgsmenc->srcpad); + gst_pad_set_chain_function (rtpgsmenc->sinkpad, gst_rtpgsmenc_chain); + gst_pad_set_link_function (rtpgsmenc->sinkpad, gst_rtpgsmenc_sinkconnect); + + rtpgsmenc->frequency = 8000; + + rtpgsmenc->next_time = 0; + rtpgsmenc->time_interval = 0; + + rtpgsmenc->seq = 0; + rtpgsmenc->ssrc = random (); +} + +static GstPadLinkReturn +gst_rtpgsmenc_sinkconnect (GstPad * pad, GstCaps * caps) +{ + GstRtpGSMEnc *rtpgsmenc; + + rtpgsmenc = GST_RTP_GSM_ENC (gst_pad_get_parent (pad)); + + gst_caps_get_int (caps, "rate", &rtpgsmenc->frequency); + + /* Pre-calculate what we can */ + rtpgsmenc->time_interval = GST_SECOND / (2 * rtpgsmenc->frequency); + + return GST_PAD_LINK_OK; +} + + +void +gst_rtpgsmenc_htons (GstBuffer *buf) +{ + gint16 *i, *len; + + /* FIXME: is this code correct or even sane at all? */ + i = (gint16 *) GST_BUFFER_DATA(buf); + len = i + GST_BUFFER_SIZE (buf) / sizeof (gint16 *); + + for (; inext_time = 0; + gst_pad_event_default (pad, event); + return; + default: + gst_pad_event_default (pad, event); + return; + } + } + + /* We only need the header */ + packet = rtp_packet_new_allocate (0, 0, 0); + + rtp_packet_set_csrc_count (packet, 0); + rtp_packet_set_extension (packet, 0); + rtp_packet_set_padding (packet, 0); + rtp_packet_set_version (packet, RTP_VERSION); + rtp_packet_set_marker (packet, 0); + rtp_packet_set_ssrc (packet, g_htonl (rtpgsmenc->ssrc)); + rtp_packet_set_seq (packet, g_htons (rtpgsmenc->seq)); + rtp_packet_set_timestamp (packet, g_htonl ((guint32) rtpgsmenc->next_time / GST_SECOND)); + rtp_packet_set_payload_type (packet, (guint8) PAYLOAD_GSM); + + /* FIXME: According to RFC 1890, this is required, right? */ +#if G_BYTE_ORDER == G_LITTLE_ENDIAN + gst_rtpgsmenc_htons (buf); +#endif + + outbuf = gst_buffer_new (); + GST_BUFFER_SIZE (outbuf) = rtp_packet_get_packet_len (packet) + GST_BUFFER_SIZE (buf); + GST_BUFFER_DATA (outbuf) = g_malloc (GST_BUFFER_SIZE (outbuf)); + GST_BUFFER_TIMESTAMP (outbuf) = rtpgsmenc->next_time; + + memcpy (GST_BUFFER_DATA (outbuf), packet->data, rtp_packet_get_packet_len (packet)); + memcpy (GST_BUFFER_DATA (outbuf) + rtp_packet_get_packet_len(packet), GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); + + GST_DEBUG (0,"gst_rtpgsmenc_chain: pushing buffer of size %d", GST_BUFFER_SIZE(outbuf)); + gst_pad_push (rtpgsmenc->srcpad, outbuf); + + ++rtpgsmenc->seq; + rtpgsmenc->next_time += rtpgsmenc->time_interval * GST_BUFFER_SIZE (buf); + + rtp_packet_free (packet); + gst_buffer_unref (buf); +} + +static void +gst_rtpgsmenc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) +{ + GstRtpGSMEnc *rtpgsmenc; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_RTP_GSM_ENC (object)); + rtpgsmenc = GST_RTP_GSM_ENC (object); + + switch (prop_id) { + default: + break; + } +} + +static void +gst_rtpgsmenc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) +{ + GstRtpGSMEnc *rtpgsmenc; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_RTP_GSM_ENC (object)); + rtpgsmenc = GST_RTP_GSM_ENC (object); + + switch (prop_id) { + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstElementStateReturn +gst_rtpgsmenc_change_state (GstElement * element) +{ + GstRtpGSMEnc *rtpgsmenc; + + g_return_val_if_fail (GST_IS_RTP_GSM_ENC (element), GST_STATE_FAILURE); + + rtpgsmenc = GST_RTP_GSM_ENC (element); + + GST_DEBUG (0, "state pending %d\n", GST_STATE_PENDING (element)); + + /* if going down into NULL state, close the file if it's open */ + switch (GST_STATE_TRANSITION (element)) { + case GST_STATE_NULL_TO_READY: + break; + + case GST_STATE_READY_TO_NULL: + break; + + default: + break; + } + + /* if we haven't failed already, give the parent class a chance to ;-) */ + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + + return GST_STATE_SUCCESS; +} + +gboolean +gst_rtpgsmenc_plugin_init (GModule * module, GstPlugin * plugin) +{ + GstElementFactory *rtpgsmenc; + + rtpgsmenc = gst_element_factory_new ("rtpgsmenc", GST_TYPE_RTP_GSM_ENC, &gst_rtpgsmenc_details); + g_return_val_if_fail (rtpgsmenc != NULL, FALSE); + + gst_element_factory_add_pad_template (rtpgsmenc, GST_PAD_TEMPLATE_GET (sink_factory)); + gst_element_factory_add_pad_template (rtpgsmenc, GST_PAD_TEMPLATE_GET (src_factory)); + + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (rtpgsmenc)); + + return TRUE; +} diff --git a/gst/rtp/gstrtpgsmpay.h b/gst/rtp/gstrtpgsmpay.h new file mode 100644 index 0000000000..76a0ec0e6d --- /dev/null +++ b/gst/rtp/gstrtpgsmpay.h @@ -0,0 +1,79 @@ +/* Gnome-Streamer + * Copyright (C) <1999> Erik Walthinsen + * + * 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_GSM_ENC_H__ +#define __GST_RTP_GSM_ENC_H__ + +#include +#include "rtp-packet.h" +#include "gstrtp-common.h" + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + +/* Definition of structure storing data for this element. */ +typedef struct _GstRtpGSMEnc GstRtpGSMEnc; +struct _GstRtpGSMEnc +{ + GstElement element; + + GstPad *sinkpad; + GstPad *srcpad; + + guint frequency; + + /* the timestamp of the next frame */ + guint64 next_time; + /* the interval between frames */ + guint64 time_interval; + + guint32 ssrc; + guint16 seq; +}; + +/* Standard definition defining a class for this element. */ +typedef struct _GstRtpGSMEncClass GstRtpGSMEncClass; +struct _GstRtpGSMEncClass +{ + GstElementClass parent_class; +}; + +/* Standard macros for defining types for this element. */ +#define GST_TYPE_RTP_GSM_ENC \ + (gst_rtpgsmenc_get_type()) +#define GST_RTP_GSM_ENC(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_GSM_ENC,GstRtpGSMEnc)) +#define GST_RTP_GSM_ENC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_GSM_ENC,GstRtpGSMEnc)) +#define GST_IS_RTP_GSM_ENC(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_GSM_ENC)) +#define GST_IS_RTP_GSM_ENC_CLASS(obj) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_GSM_ENC)) + +gboolean gst_rtpgsmenc_plugin_init (GModule * module, GstPlugin * plugin); + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + + +#endif /* __GST_RTP_GSM_ENC_H__ */ diff --git a/gst/rtp/rtp-packet.c b/gst/rtp/rtp-packet.c index a6fca8b2e0..0564fd944c 100644 --- a/gst/rtp/rtp-packet.c +++ b/gst/rtp/rtp-packet.c @@ -93,7 +93,7 @@ rtp_packet_free(Rtp_Packet packet) g_free(packet); } -Rtp_Packet +/*Rtp_Packet rtp_packet_read(int fd, struct sockaddr *fromaddr, socklen_t *fromlen) { int packlen; @@ -105,14 +105,14 @@ rtp_packet_read(int fd, struct sockaddr *fromaddr, socklen_t *fromlen) if (packlen < 0) { g_error("rtp_packet_read: recvfrom: %d %s", errno, strerror(errno)); - /*exit(1);*/ + //exit(1); return NULL; } return rtp_packet_new_take_data(buf, packlen); -} +}*/ -void +/*void rtp_packet_send(Rtp_Packet packet, int fd, struct sockaddr *toaddr, socklen_t tolen) { g_return_if_fail(packet != NULL); @@ -120,7 +120,7 @@ rtp_packet_send(Rtp_Packet packet, int fd, struct sockaddr *toaddr, socklen_t to sendto(fd, (void *) packet -> data, packet -> data_len, 0, toaddr, tolen); -} +}*/ guint8 rtp_packet_get_version(Rtp_Packet packet) diff --git a/gst/rtp/rtp-packet.h b/gst/rtp/rtp-packet.h index f19f8c6ea2..f731c5f0fc 100644 --- a/gst/rtp/rtp-packet.h +++ b/gst/rtp/rtp-packet.h @@ -22,9 +22,7 @@ #ifndef _RTP_PACKET_H #define _RTP_PACKET_H 1 -#include #include -#include #include #ifdef __sun @@ -35,11 +33,9 @@ extern "C" { #endif -enum { - RTP_VERSION = 2, - RTP_HEADER_LEN = 12, - RTP_MTU = 2048 -}; +#define RTP_VERSION 2 +#define RTP_HEADER_LEN 12 +#define RTP_MTU 2048 typedef struct Rtp_Header *Rtp_Header; @@ -79,8 +75,8 @@ Rtp_Packet rtp_packet_new_copy_data(gpointer data, guint data_len); Rtp_Packet rtp_packet_new_allocate(guint payload_len, guint pad_len, guint csrc_count); void rtp_packet_free(Rtp_Packet packet); -Rtp_Packet rtp_packet_read(int fd, struct sockaddr *fromaddr, socklen_t *fromlen); -void rtp_packet_send(Rtp_Packet packet, int fd, struct sockaddr *toaddr, socklen_t tolen); +//Rtp_Packet rtp_packet_read(int fd, struct sockaddr *fromaddr, socklen_t *fromlen); +//void rtp_packet_send(Rtp_Packet packet, int fd, struct sockaddr *toaddr, socklen_t tolen); guint8 rtp_packet_get_version(Rtp_Packet packet); void rtp_packet_set_version(Rtp_Packet packet, guint8 version); guint8 rtp_packet_get_padding(Rtp_Packet packet);