diff options
author | Zeeshan Ali <zeenix@gmail.com> | 2003-07-13 00:20:44 +0000 |
---|---|---|
committer | Zeeshan Ali <zeenix@gmail.com> | 2003-07-13 00:20:44 +0000 |
commit | 333468b5d448062b40c428e5c3b95dcb8159807c (patch) | |
tree | ddaa5d09eae9309ef88260439ef4379d26ed9af8 /gst/rtp/gstrtpgsmpay.c | |
parent | 8c4aec99981a02e6fe6fdf61d4156746131a442d (diff) |
GSM -> RTP and viceversa
Original commit message from CVS:
GSM -> RTP and viceversa
Diffstat (limited to 'gst/rtp/gstrtpgsmpay.c')
-rw-r--r-- | gst/rtp/gstrtpgsmpay.c | 310 |
1 files changed, 310 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpgsmpay.c b/gst/rtp/gstrtpgsmpay.c new file mode 100644 index 00000000..2fff85c6 --- /dev/null +++ b/gst/rtp/gstrtpgsmpay.c @@ -0,0 +1,310 @@ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> + * + * 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 <string.h> +#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 <zak147@yahoo.com>", + "(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 (; i<len; i++) { + *i = g_htons (*i); + } +} + +static void +gst_rtpgsmenc_chain (GstPad * pad, GstBuffer * buf) +{ + GstRtpGSMEnc *rtpgsmenc; + GstBuffer *outbuf; + Rtp_Packet packet; + + g_return_if_fail (pad != NULL); + g_return_if_fail (GST_IS_PAD (pad)); + g_return_if_fail (buf != NULL); + + rtpgsmenc = GST_RTP_GSM_ENC (GST_OBJECT_PARENT (pad)); + + g_return_if_fail (rtpgsmenc != NULL); + g_return_if_fail (GST_IS_RTP_GSM_ENC (rtpgsmenc)); + + if (GST_IS_EVENT (buf)) { + GstEvent *event = GST_EVENT (buf); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_DISCONTINUOUS: + GST_DEBUG (GST_CAT_EVENT, "discont"); + rtpgsmenc->next_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; +} |