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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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80f8780e92
Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
221 lines
5.9 KiB
C
221 lines
5.9 KiB
C
/* GStreamer
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/audio/audio.h>
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#include <stdlib.h>
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#include <string.h>
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#include "gstrtpelements.h"
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#include "gstrtpg729depay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpg729depay_debug);
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#define GST_CAT_DEFAULT (rtpg729depay_debug)
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/* references:
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*
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* RFC 3551 (4.5.6)
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*/
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0
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};
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/* input is an RTP packet
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*
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*/
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static GstStaticPadTemplate gst_rtp_g729_depay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) \"G729\"; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
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"clock-rate = (int) 8000")
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);
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static GstStaticPadTemplate gst_rtp_g729_depay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/G729, " "channels = (int) 1," "rate = (int) 8000")
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);
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static gboolean gst_rtp_g729_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_g729_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp);
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#define gst_rtp_g729_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpG729Depay, gst_rtp_g729_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpg729depay, "rtpg729depay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_G729_DEPAY, rtp_element_init (plugin));
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static void
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gst_rtp_g729_depay_class_init (GstRtpG729DepayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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GST_DEBUG_CATEGORY_INIT (rtpg729depay_debug, "rtpg729depay", 0,
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"G.729 RTP Depayloader");
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_g729_depay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_g729_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP G.729 depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts G.729 audio from RTP packets (RFC 3551)",
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"Laurent Glayal <spglegle@yahoo.fr>");
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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_g729_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_g729_depay_setcaps;
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}
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static void
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gst_rtp_g729_depay_init (GstRtpG729Depay * rtpg729depay)
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{
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GstRTPBaseDepayload *depayload;
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depayload = GST_RTP_BASE_DEPAYLOAD (rtpg729depay);
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gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
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}
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static gboolean
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gst_rtp_g729_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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GstCaps *srccaps;
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GstRtpG729Depay *rtpg729depay;
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const gchar *params;
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gint clock_rate, channels;
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gboolean ret;
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rtpg729depay = GST_RTP_G729_DEPAY (depayload);
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structure = gst_caps_get_structure (caps, 0);
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if (!(params = gst_structure_get_string (structure, "encoding-params")))
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channels = 1;
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else {
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channels = atoi (params);
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}
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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clock_rate = 8000;
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if (channels != 1)
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goto wrong_channels;
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if (clock_rate != 8000)
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goto wrong_clock_rate;
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depayload->clock_rate = clock_rate;
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srccaps = gst_caps_new_simple ("audio/G729",
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"channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, clock_rate, NULL);
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ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
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return ret;
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/* ERRORS */
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wrong_channels:
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{
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GST_DEBUG_OBJECT (rtpg729depay, "expected 1 channel, got %d", channels);
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return FALSE;
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}
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wrong_clock_rate:
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{
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GST_DEBUG_OBJECT (rtpg729depay, "expected 8000 clock-rate, got %d",
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clock_rate);
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return FALSE;
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}
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}
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static GstBuffer *
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gst_rtp_g729_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
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GstRtpG729Depay *rtpg729depay;
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GstBuffer *outbuf = NULL;
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gint payload_len;
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gboolean marker;
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rtpg729depay = GST_RTP_G729_DEPAY (depayload);
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payload_len = gst_rtp_buffer_get_payload_len (rtp);
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/* At least 2 bytes (CNG from G729 Annex B) */
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if (payload_len < 2) {
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GST_ELEMENT_WARNING (rtpg729depay, STREAM, DECODE,
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(NULL), ("G729 RTP payload too small (%d)", payload_len));
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goto bad_packet;
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}
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GST_LOG_OBJECT (rtpg729depay, "payload len %d", payload_len);
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if ((payload_len % 10) == 2) {
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GST_LOG_OBJECT (rtpg729depay, "G729 payload contains CNG frame");
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}
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outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
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marker = gst_rtp_buffer_get_marker (rtp);
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if (marker) {
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/* marker bit starts talkspurt */
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
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}
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gst_rtp_drop_non_audio_meta (depayload, outbuf);
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GST_LOG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
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gst_buffer_get_size (outbuf));
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return outbuf;
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/* ERRORS */
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bad_packet:
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{
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/* no fatal error */
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return NULL;
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}
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}
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