mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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b0afaffc5d
This allows downstream of a payloader to know the RTP header's marker flag without first having to map the buffer and parse the RTP header. Especially inside RTP header extension implementations this can be useful to decide which packet corresponds to e.g. the last packet of a video frame. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1776>
305 lines
8.7 KiB
C
305 lines
8.7 KiB
C
/* GStreamer
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* Copyright (C) <2007> Nokia Corporation
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* Copyright (C) <2007> Collabora Ltd
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* @author: Olivier Crete <olivier.crete@collabora.co.uk>
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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 <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/base/gstadapter.h>
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#include <gst/audio/audio.h>
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#include "gstrtpelements.h"
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#include "gstrtpg723pay.h"
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#include "gstrtputils.h"
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#define G723_FRAME_DURATION (30 * GST_MSECOND)
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static gboolean gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_g723_pay_handle_buffer (GstRTPBasePayload *
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payload, GstBuffer * buf);
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static GstStaticPadTemplate gst_rtp_g723_pay_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 ("audio/G723, " /* according to RFC 3551 */
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"channels = (int) 1, " "rate = (int) 8000")
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);
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static GstStaticPadTemplate gst_rtp_g723_pay_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 ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_G723_STRING ", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) \"G723\"; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"G723\"")
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);
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static void gst_rtp_g723_pay_finalize (GObject * object);
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static GstStateChangeReturn gst_rtp_g723_pay_change_state (GstElement * element,
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GstStateChange transition);
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#define gst_rtp_g723_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRTPG723Pay, gst_rtp_g723_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpg723pay, "rtpg723pay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_G723_PAY, rtp_element_init (plugin));
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static void
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gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *payload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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payload_class = (GstRTPBasePayloadClass *) klass;
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gobject_class->finalize = gst_rtp_g723_pay_finalize;
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gstelement_class->change_state = gst_rtp_g723_pay_change_state;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_g723_pay_sink_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_g723_pay_src_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP G.723 payloader", "Codec/Payloader/Network/RTP",
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"Packetize G.723 audio into RTP packets",
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"Wim Taymans <wim.taymans@gmail.com>");
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payload_class->set_caps = gst_rtp_g723_pay_set_caps;
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payload_class->handle_buffer = gst_rtp_g723_pay_handle_buffer;
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}
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static void
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gst_rtp_g723_pay_init (GstRTPG723Pay * pay)
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{
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GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
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pay->adapter = gst_adapter_new ();
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payload->pt = GST_RTP_PAYLOAD_G723;
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}
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static void
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gst_rtp_g723_pay_finalize (GObject * object)
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{
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GstRTPG723Pay *pay;
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pay = GST_RTP_G723_PAY (object);
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g_object_unref (pay->adapter);
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pay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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gboolean res;
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gst_rtp_base_payload_set_options (payload, "audio",
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payload->pt != GST_RTP_PAYLOAD_G723, "G723", 8000);
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res = gst_rtp_base_payload_set_outcaps (payload, NULL);
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return res;
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}
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static GstFlowReturn
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gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
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{
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GstBuffer *outbuf, *payload_buf;
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GstFlowReturn ret;
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guint avail;
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GstRTPBuffer rtp = { NULL };
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avail = gst_adapter_available (pay->adapter);
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outbuf =
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gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD (pay),
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0, 0, 0);
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gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
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GST_BUFFER_PTS (outbuf) = pay->timestamp;
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GST_BUFFER_DURATION (outbuf) = pay->duration;
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/* copy G723 data as payload */
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payload_buf = gst_adapter_take_buffer_fast (pay->adapter, avail);
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pay->timestamp = GST_CLOCK_TIME_NONE;
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pay->duration = 0;
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/* set discont and marker */
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if (pay->discont) {
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MARKER);
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gst_rtp_buffer_set_marker (&rtp, TRUE);
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pay->discont = FALSE;
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}
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gst_rtp_buffer_unmap (&rtp);
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gst_rtp_copy_audio_meta (pay, outbuf, payload_buf);
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outbuf = gst_buffer_append (outbuf, payload_buf);
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ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (pay), outbuf);
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return ret;
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}
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/* 00 high-rate speech (6.3 kb/s) 24
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* 01 low-rate speech (5.3 kb/s) 20
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* 10 SID frame 4
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* 11 reserved 0 */
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static const guint size_tab[4] = {
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24, 20, 4, 0
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};
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static GstFlowReturn
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gst_rtp_g723_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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GstMapInfo map;
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guint8 HDR;
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GstRTPG723Pay *pay;
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GstClockTime packet_dur, timestamp;
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guint payload_len, packet_len;
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pay = GST_RTP_G723_PAY (payload);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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timestamp = GST_BUFFER_PTS (buf);
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if (GST_BUFFER_IS_DISCONT (buf)) {
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/* flush everything on discont */
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gst_adapter_clear (pay->adapter);
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pay->timestamp = GST_CLOCK_TIME_NONE;
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pay->duration = 0;
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pay->discont = TRUE;
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}
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/* should be one of these sizes */
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if (map.size != 4 && map.size != 20 && map.size != 24)
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goto invalid_size;
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/* check size by looking at the header bits */
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HDR = map.data[0] & 0x3;
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if (size_tab[HDR] != map.size)
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goto wrong_size;
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/* calculate packet size and duration */
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payload_len = gst_adapter_available (pay->adapter) + map.size;
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packet_dur = pay->duration + G723_FRAME_DURATION;
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packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
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if (gst_rtp_base_payload_is_filled (payload, packet_len, packet_dur)) {
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/* size or duration would overflow the packet, flush the queued data */
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ret = gst_rtp_g723_pay_flush (pay);
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}
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/* update timestamp, we keep the timestamp for the first packet in the adapter
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* but are able to calculate it from next packets. */
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if (timestamp != GST_CLOCK_TIME_NONE && pay->timestamp == GST_CLOCK_TIME_NONE) {
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if (timestamp > pay->duration)
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pay->timestamp = timestamp - pay->duration;
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else
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pay->timestamp = 0;
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}
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gst_buffer_unmap (buf, &map);
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/* add packet to the queue */
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gst_adapter_push (pay->adapter, buf);
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pay->duration = packet_dur;
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/* check if we can flush now */
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if (pay->duration >= payload->min_ptime) {
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ret = gst_rtp_g723_pay_flush (pay);
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}
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return ret;
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/* WARNINGS */
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invalid_size:
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{
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GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
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("Invalid input buffer size"),
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("Input size should be 4, 20 or 24, got %" G_GSIZE_FORMAT, map.size));
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gst_buffer_unmap (buf, &map);
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gst_buffer_unref (buf);
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return GST_FLOW_OK;
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}
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wrong_size:
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{
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GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
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("Wrong input buffer size"),
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("Expected input buffer size %u but got %" G_GSIZE_FORMAT,
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size_tab[HDR], map.size));
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gst_buffer_unmap (buf, &map);
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gst_buffer_unref (buf);
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return GST_FLOW_OK;
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}
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}
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static GstStateChangeReturn
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gst_rtp_g723_pay_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret;
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GstRTPG723Pay *pay;
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pay = GST_RTP_G723_PAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_adapter_clear (pay->adapter);
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pay->timestamp = GST_CLOCK_TIME_NONE;
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pay->duration = 0;
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pay->discont = TRUE;
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_adapter_clear (pay->adapter);
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break;
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default:
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break;
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}
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return ret;
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}
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