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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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This is required to add RTP header extensions from the meta automatically. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/674>
479 lines
13 KiB
C
479 lines
13 KiB
C
/* GStreamer
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* Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.com>
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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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/**
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* SECTION:element-rtpac3pay
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* @title: rtpac3pay
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* @see_also: rtpac3depay
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*
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* Payload AC3 audio into RTP packets according to RFC 4184.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt
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*
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* ## Example pipeline
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* |[
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* gst-launch-1.0 -v audiotestsrc ! avenc_ac3 ! rtpac3pay ! udpsink
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* ]| This example pipeline will encode and payload AC3 stream. Refer to
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* the rtpac3depay example to depayload and decode the RTP stream.
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*
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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/audio/audio.h>
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#include "gstrtpac3pay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpac3pay_debug);
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#define GST_CAT_DEFAULT (rtpac3pay_debug)
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static GstStaticPadTemplate gst_rtp_ac3_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/ac3; " "audio/x-ac3; ")
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);
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static GstStaticPadTemplate gst_rtp_ac3_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_DYNAMIC_STRING ", "
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"clock-rate = (int) { 32000, 44100, 48000 }, "
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"encoding-name = (string) \"AC3\"")
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);
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static void gst_rtp_ac3_pay_finalize (GObject * object);
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static GstStateChangeReturn gst_rtp_ac3_pay_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_rtp_ac3_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static gboolean gst_rtp_ac3_pay_sink_event (GstRTPBasePayload * payload,
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GstEvent * event);
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static GstFlowReturn gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay);
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static GstFlowReturn gst_rtp_ac3_pay_handle_buffer (GstRTPBasePayload * payload,
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GstBuffer * buffer);
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#define gst_rtp_ac3_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpAC3Pay, gst_rtp_ac3_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static void
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gst_rtp_ac3_pay_class_init (GstRtpAC3PayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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GST_DEBUG_CATEGORY_INIT (rtpac3pay_debug, "rtpac3pay", 0,
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"AC3 Audio RTP Depayloader");
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gobject_class->finalize = gst_rtp_ac3_pay_finalize;
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gstelement_class->change_state = gst_rtp_ac3_pay_change_state;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_ac3_pay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_ac3_pay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP AC3 audio payloader", "Codec/Payloader/Network/RTP",
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"Payload AC3 audio as RTP packets (RFC 4184)",
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"Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasepayload_class->set_caps = gst_rtp_ac3_pay_setcaps;
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gstrtpbasepayload_class->sink_event = gst_rtp_ac3_pay_sink_event;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_ac3_pay_handle_buffer;
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}
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static void
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gst_rtp_ac3_pay_init (GstRtpAC3Pay * rtpac3pay)
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{
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rtpac3pay->adapter = gst_adapter_new ();
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}
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static void
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gst_rtp_ac3_pay_finalize (GObject * object)
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{
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GstRtpAC3Pay *rtpac3pay;
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rtpac3pay = GST_RTP_AC3_PAY (object);
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g_object_unref (rtpac3pay->adapter);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_rtp_ac3_pay_reset (GstRtpAC3Pay * pay)
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{
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pay->first_ts = -1;
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pay->duration = 0;
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gst_adapter_clear (pay->adapter);
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GST_DEBUG_OBJECT (pay, "reset depayloader");
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}
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static gboolean
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gst_rtp_ac3_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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gboolean res;
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gint rate;
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GstStructure *structure;
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "rate", &rate))
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rate = 90000; /* default */
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gst_rtp_base_payload_set_options (payload, "audio", TRUE, "AC3", rate);
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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 gboolean
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gst_rtp_ac3_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
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{
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gboolean res;
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GstRtpAC3Pay *rtpac3pay;
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rtpac3pay = GST_RTP_AC3_PAY (payload);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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/* make sure we push the last packets in the adapter on EOS */
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gst_rtp_ac3_pay_flush (rtpac3pay);
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break;
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case GST_EVENT_FLUSH_STOP:
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gst_rtp_ac3_pay_reset (rtpac3pay);
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break;
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default:
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break;
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}
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res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
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return res;
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}
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struct frmsize_s
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{
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guint16 bit_rate;
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guint16 frm_size[3];
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};
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static const struct frmsize_s frmsizecod_tbl[] = {
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{32, {64, 69, 96}},
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{32, {64, 70, 96}},
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{40, {80, 87, 120}},
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{40, {80, 88, 120}},
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{48, {96, 104, 144}},
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{48, {96, 105, 144}},
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{56, {112, 121, 168}},
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{56, {112, 122, 168}},
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{64, {128, 139, 192}},
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{64, {128, 140, 192}},
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{80, {160, 174, 240}},
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{80, {160, 175, 240}},
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{96, {192, 208, 288}},
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{96, {192, 209, 288}},
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{112, {224, 243, 336}},
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{112, {224, 244, 336}},
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{128, {256, 278, 384}},
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{128, {256, 279, 384}},
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{160, {320, 348, 480}},
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{160, {320, 349, 480}},
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{192, {384, 417, 576}},
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{192, {384, 418, 576}},
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{224, {448, 487, 672}},
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{224, {448, 488, 672}},
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{256, {512, 557, 768}},
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{256, {512, 558, 768}},
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{320, {640, 696, 960}},
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{320, {640, 697, 960}},
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{384, {768, 835, 1152}},
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{384, {768, 836, 1152}},
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{448, {896, 975, 1344}},
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{448, {896, 976, 1344}},
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{512, {1024, 1114, 1536}},
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{512, {1024, 1115, 1536}},
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{576, {1152, 1253, 1728}},
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{576, {1152, 1254, 1728}},
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{640, {1280, 1393, 1920}},
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{640, {1280, 1394, 1920}}
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};
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static GstFlowReturn
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gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay)
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{
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guint avail, FT, NF, mtu;
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GstBuffer *outbuf;
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GstFlowReturn ret;
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/* the data available in the adapter is either smaller
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* than the MTU or bigger. In the case it is smaller, the complete
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* adapter contents can be put in one packet. In the case the
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* adapter has more than one MTU, we need to split the AC3 data
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* over multiple packets. */
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avail = gst_adapter_available (rtpac3pay->adapter);
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ret = GST_FLOW_OK;
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FT = 0;
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/* number of frames */
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NF = rtpac3pay->NF;
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mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpac3pay);
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GST_LOG_OBJECT (rtpac3pay, "flushing %u bytes", avail);
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while (avail > 0) {
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guint towrite;
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guint8 *payload;
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guint payload_len;
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guint packet_len;
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GstRTPBuffer rtp = { NULL, };
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GstBuffer *payload_buffer;
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/* this will be the total length of the packet */
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packet_len = gst_rtp_buffer_calc_packet_len (2 + avail, 0, 0);
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/* fill one MTU or all available bytes */
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towrite = MIN (packet_len, mtu);
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/* this is the payload length */
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payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
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/* create buffer to hold the payload */
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outbuf =
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gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
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(rtpac3pay), 2, 0, 0);
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if (FT == 0) {
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/* check if it all fits */
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if (towrite < packet_len) {
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guint maxlen;
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GST_LOG_OBJECT (rtpac3pay, "we need to fragment");
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/* check if we will be able to put at least 5/8th of the total
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* frame in this first frame. */
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if ((avail * 5) / 8 >= (payload_len - 2))
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FT = 1;
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else
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FT = 2;
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/* check how many fragments we will need */
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maxlen = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
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NF = (avail + maxlen - 1) / maxlen;
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}
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} else if (FT != 3) {
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/* remaining fragment */
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FT = 3;
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}
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/*
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* 0 1
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | MBZ | FT| NF |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*
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* FT: 0: one or more complete frames
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* 1: initial 5/8 fragment
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* 2: initial fragment not 5/8
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* 3: other fragment
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* NF: amount of frames if FT = 0, else number of fragments.
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*/
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gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
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GST_LOG_OBJECT (rtpac3pay, "FT %u, NF %u", FT, NF);
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payload = gst_rtp_buffer_get_payload (&rtp);
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payload[0] = (FT & 3);
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payload[1] = NF;
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payload_len -= 2;
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if (avail == payload_len)
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gst_rtp_buffer_set_marker (&rtp, TRUE);
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gst_rtp_buffer_unmap (&rtp);
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payload_buffer =
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gst_adapter_take_buffer_fast (rtpac3pay->adapter, payload_len);
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gst_rtp_copy_audio_meta (rtpac3pay, outbuf, payload_buffer);
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outbuf = gst_buffer_append (outbuf, payload_buffer);
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avail -= payload_len;
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GST_BUFFER_PTS (outbuf) = rtpac3pay->first_ts;
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GST_BUFFER_DURATION (outbuf) = rtpac3pay->duration;
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ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpac3pay), outbuf);
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}
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return ret;
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}
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static GstFlowReturn
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gst_rtp_ac3_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpAC3Pay *rtpac3pay;
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GstFlowReturn ret;
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gsize avail, left, NF;
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GstMapInfo map;
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guint8 *p;
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guint packet_len;
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GstClockTime duration, timestamp;
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rtpac3pay = GST_RTP_AC3_PAY (basepayload);
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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duration = GST_BUFFER_DURATION (buffer);
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timestamp = GST_BUFFER_PTS (buffer);
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if (GST_BUFFER_IS_DISCONT (buffer)) {
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GST_DEBUG_OBJECT (rtpac3pay, "DISCONT");
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gst_rtp_ac3_pay_reset (rtpac3pay);
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}
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/* count the amount of incoming packets */
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NF = 0;
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left = map.size;
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p = map.data;
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while (TRUE) {
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guint bsid, fscod, frmsizecod, frame_size;
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if (left < 6)
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break;
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if (p[0] != 0x0b || p[1] != 0x77)
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break;
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bsid = p[5] >> 3;
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if (bsid > 8)
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break;
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frmsizecod = p[4] & 0x3f;
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fscod = p[4] >> 6;
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GST_DEBUG_OBJECT (rtpac3pay, "fscod %u, %u", fscod, frmsizecod);
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if (fscod >= 3 || frmsizecod >= 38)
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break;
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frame_size = frmsizecod_tbl[frmsizecod].frm_size[fscod] * 2;
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if (frame_size > left)
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break;
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NF++;
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GST_DEBUG_OBJECT (rtpac3pay, "found frame %" G_GSIZE_FORMAT " of size %u",
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NF, frame_size);
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p += frame_size;
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left -= frame_size;
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}
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gst_buffer_unmap (buffer, &map);
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if (NF == 0)
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goto no_frames;
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avail = gst_adapter_available (rtpac3pay->adapter);
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/* get packet length of previous data and this new data,
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* payload length includes a 4 byte header */
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packet_len = gst_rtp_buffer_calc_packet_len (2 + avail + map.size, 0, 0);
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/* if this buffer is going to overflow the packet, flush what we
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* have. */
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if (gst_rtp_base_payload_is_filled (basepayload,
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packet_len, rtpac3pay->duration + duration)) {
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ret = gst_rtp_ac3_pay_flush (rtpac3pay);
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avail = 0;
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} else {
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ret = GST_FLOW_OK;
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}
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if (avail == 0) {
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GST_DEBUG_OBJECT (rtpac3pay,
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"first packet, save timestamp %" GST_TIME_FORMAT,
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GST_TIME_ARGS (timestamp));
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rtpac3pay->first_ts = timestamp;
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rtpac3pay->duration = 0;
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rtpac3pay->NF = 0;
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}
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gst_adapter_push (rtpac3pay->adapter, buffer);
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rtpac3pay->duration += duration;
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rtpac3pay->NF += NF;
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return ret;
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/* ERRORS */
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no_frames:
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{
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GST_WARNING_OBJECT (rtpac3pay, "no valid AC3 frames found");
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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_ac3_pay_change_state (GstElement * element, GstStateChange transition)
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{
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GstRtpAC3Pay *rtpac3pay;
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GstStateChangeReturn ret;
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rtpac3pay = GST_RTP_AC3_PAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_rtp_ac3_pay_reset (rtpac3pay);
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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_rtp_ac3_pay_reset (rtpac3pay);
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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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gboolean
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gst_rtp_ac3_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpac3pay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_AC3_PAY);
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}
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