/* GStreamer * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include <gst/rtp/gstrtpbuffer.h> #include <gst/audio/audio.h> #include <stdlib.h> #include <string.h> #include "gstrtpelements.h" #include "gstrtpg729depay.h" #include "gstrtputils.h" GST_DEBUG_CATEGORY_STATIC (rtpg729depay_debug); #define GST_CAT_DEFAULT (rtpg729depay_debug) /* references: * * RFC 3551 (4.5.6) */ enum { /* FILL ME */ LAST_SIGNAL }; enum { PROP_0 }; /* input is an RTP packet * */ static GstStaticPadTemplate gst_rtp_g729_depay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"; " "application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", " "clock-rate = (int) 8000") ); static GstStaticPadTemplate gst_rtp_g729_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/G729, " "channels = (int) 1," "rate = (int) 8000") ); static gboolean gst_rtp_g729_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_g729_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp); #define gst_rtp_g729_depay_parent_class parent_class G_DEFINE_TYPE (GstRtpG729Depay, gst_rtp_g729_depay, GST_TYPE_RTP_BASE_DEPAYLOAD); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpg729depay, "rtpg729depay", GST_RANK_SECONDARY, GST_TYPE_RTP_G729_DEPAY, rtp_element_init (plugin)); static void gst_rtp_g729_depay_class_init (GstRtpG729DepayClass * klass) { GstElementClass *gstelement_class; GstRTPBaseDepayloadClass *gstrtpbasedepayload_class; GST_DEBUG_CATEGORY_INIT (rtpg729depay_debug, "rtpg729depay", 0, "G.729 RTP Depayloader"); gstelement_class = (GstElementClass *) klass; gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass; gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_g729_depay_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_g729_depay_sink_template); gst_element_class_set_static_metadata (gstelement_class, "RTP G.729 depayloader", "Codec/Depayloader/Network/RTP", "Extracts G.729 audio from RTP packets (RFC 3551)", "Laurent Glayal <spglegle@yahoo.fr>"); gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_g729_depay_process; gstrtpbasedepayload_class->set_caps = gst_rtp_g729_depay_setcaps; } static void gst_rtp_g729_depay_init (GstRtpG729Depay * rtpg729depay) { GstRTPBaseDepayload *depayload; depayload = GST_RTP_BASE_DEPAYLOAD (rtpg729depay); gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload)); } static gboolean gst_rtp_g729_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) { GstStructure *structure; GstCaps *srccaps; GstRtpG729Depay *rtpg729depay; const gchar *params; gint clock_rate, channels; gboolean ret; rtpg729depay = GST_RTP_G729_DEPAY (depayload); structure = gst_caps_get_structure (caps, 0); if (!(params = gst_structure_get_string (structure, "encoding-params"))) channels = 1; else { channels = atoi (params); } if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) clock_rate = 8000; if (channels != 1) goto wrong_channels; if (clock_rate != 8000) goto wrong_clock_rate; depayload->clock_rate = clock_rate; srccaps = gst_caps_new_simple ("audio/G729", "channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, clock_rate, NULL); ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps); gst_caps_unref (srccaps); return ret; /* ERRORS */ wrong_channels: { GST_DEBUG_OBJECT (rtpg729depay, "expected 1 channel, got %d", channels); return FALSE; } wrong_clock_rate: { GST_DEBUG_OBJECT (rtpg729depay, "expected 8000 clock-rate, got %d", clock_rate); return FALSE; } } static GstBuffer * gst_rtp_g729_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp) { GstRtpG729Depay *rtpg729depay; GstBuffer *outbuf = NULL; gint payload_len; gboolean marker; rtpg729depay = GST_RTP_G729_DEPAY (depayload); payload_len = gst_rtp_buffer_get_payload_len (rtp); /* At least 2 bytes (CNG from G729 Annex B) */ if (payload_len < 2) { GST_ELEMENT_WARNING (rtpg729depay, STREAM, DECODE, (NULL), ("G729 RTP payload too small (%d)", payload_len)); goto bad_packet; } GST_LOG_OBJECT (rtpg729depay, "payload len %d", payload_len); if ((payload_len % 10) == 2) { GST_LOG_OBJECT (rtpg729depay, "G729 payload contains CNG frame"); } outbuf = gst_rtp_buffer_get_payload_buffer (rtp); marker = gst_rtp_buffer_get_marker (rtp); if (marker) { /* marker bit starts talkspurt */ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC); } gst_rtp_drop_non_audio_meta (depayload, outbuf); GST_LOG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT, gst_buffer_get_size (outbuf)); return outbuf; /* ERRORS */ bad_packet: { /* no fatal error */ return NULL; } }