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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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4a28e649c3
Every g_quark_from_static_string() is a hash table lookup serialised on the global quark lock in GLib. Let's just look up the two quarks we need once and cache them locally for future use. While we're at it, add new utility functions for the two most commonly used tags (audio + video). Make first argument a gpointer so we don't have to cast and make the code ugly. These are used for logging purposes only anyway.
477 lines
13 KiB
C
477 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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* @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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* <refsect2>
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* <title>Example pipeline</title>
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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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* </refsect2>
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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 = gst_rtp_buffer_new_allocate (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 incomming 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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