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bce1d121ba
audio/x-ac3 is the canonical media format in GStreamer. audio/ac3 is sometimes accepted as input (e.g. in rtpac3pay or ac3parse), but shouldn't be output. Fixes #3038. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5472>
176 lines
5.3 KiB
C
176 lines
5.3 KiB
C
/* GStreamer
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* Copyright (C) <2007> 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-rtpac3depay
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* @title: rtpac3depay
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* @see_also: rtpac3pay
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*
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* Extract AC3 audio from 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 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)AC3, payload=(int)96' ! rtpac3depay ! a52dec ! pulsesink
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* ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to
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* the rtpac3pay example to create 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 <gst/rtp/gstrtpbuffer.h>
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#include <gst/audio/audio.h>
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#include <string.h>
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#include "gstrtpelements.h"
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#include "gstrtpac3depay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpac3depay_debug);
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#define GST_CAT_DEFAULT (rtpac3depay_debug)
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static GstStaticPadTemplate gst_rtp_ac3_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/x-ac3")
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);
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static GstStaticPadTemplate gst_rtp_ac3_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) { 32000, 44100, 48000 }, "
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"encoding-name = (string) \"AC3\"")
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);
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G_DEFINE_TYPE (GstRtpAC3Depay, gst_rtp_ac3_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpac3depay, "rtpac3depay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_AC3_DEPAY, rtp_element_init (plugin));
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static gboolean gst_rtp_ac3_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_ac3_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp);
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static void
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gst_rtp_ac3_depay_class_init (GstRtpAC3DepayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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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_ac3_depay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_ac3_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP AC3 depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts AC3 audio from RTP packets (RFC 4184)",
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"Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasedepayload_class->set_caps = gst_rtp_ac3_depay_setcaps;
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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_ac3_depay_process;
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GST_DEBUG_CATEGORY_INIT (rtpac3depay_debug, "rtpac3depay", 0,
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"AC3 Audio RTP Depayloader");
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}
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static void
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gst_rtp_ac3_depay_init (GstRtpAC3Depay * rtpac3depay)
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{
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/* needed because of G_DEFINE_TYPE */
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}
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static gboolean
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gst_rtp_ac3_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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gint clock_rate;
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GstCaps *srccaps;
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gboolean res;
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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clock_rate = 90000; /* default */
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depayload->clock_rate = clock_rate;
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srccaps = gst_caps_new_empty_simple ("audio/x-ac3");
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res = gst_pad_set_caps (depayload->srcpad, srccaps);
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gst_caps_unref (srccaps);
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return res;
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}
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static GstBuffer *
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gst_rtp_ac3_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
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GstRtpAC3Depay *rtpac3depay;
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GstBuffer *outbuf;
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guint8 *payload;
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guint16 FT, NF;
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rtpac3depay = GST_RTP_AC3_DEPAY (depayload);
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if (gst_rtp_buffer_get_payload_len (rtp) < 2)
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goto empty_packet;
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payload = gst_rtp_buffer_get_payload (rtp);
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/* strip off header
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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 = payload[0] & 0x3;
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NF = payload[1];
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GST_DEBUG_OBJECT (rtpac3depay, "FT: %d, NF: %d", FT, NF);
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/* We don't bother with fragmented packets yet */
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outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, 2, -1);
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if (outbuf) {
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gst_rtp_drop_non_audio_meta (rtpac3depay, outbuf);
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GST_DEBUG_OBJECT (rtpac3depay, "pushing buffer of size %" G_GSIZE_FORMAT,
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gst_buffer_get_size (outbuf));
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}
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return outbuf;
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/* ERRORS */
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empty_packet:
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{
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GST_ELEMENT_WARNING (rtpac3depay, STREAM, DECODE,
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("Empty Payload."), (NULL));
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return NULL;
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}
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}
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