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478 lines
14 KiB
C
478 lines
14 KiB
C
/* GStreamer
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* Copyright (C) <2005> 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-rtpamrdepay
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* @title: rtpamrdepay
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* @see_also: rtpamrpay
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*
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* Extract AMR audio from RTP packets according to RFC 3267.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.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)8000, encoding-name=(string)AMR, encoding-params=(string)1, octet-align=(string)1, payload=(int)96' ! rtpamrdepay ! amrnbdec ! pulsesink
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* ]| This example pipeline will depayload and decode an RTP AMR stream. Refer to
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* the rtpamrpay example to create the RTP stream.
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*
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*/
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/*
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* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
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* Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
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* Wideband (AMR-WB) Audio Codecs.
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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 <stdlib.h>
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#include <string.h>
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#include "gstrtpamrdepay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpamrdepay_debug);
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#define GST_CAT_DEFAULT (rtpamrdepay_debug)
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/* RtpAMRDepay signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0
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};
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/* input is an RTP packet
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*
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* params see RFC 3267, section 8.1
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*/
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static GstStaticPadTemplate gst_rtp_amr_depay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"AMR\", "
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/* This is the default, so the peer doesn't have to specify it
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* "encoding-params = (string) \"1\", " */
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/* NOTE that all values must be strings in orde to be able to do SDP <->
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* GstCaps mapping. */
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"octet-align = (string) \"1\";"
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/* following options are not needed for a decoder
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*
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"crc = (string) { \"0\", \"1\" }, "
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"robust-sorting = (string) \"0\", "
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"interleaving = (string) \"0\";"
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"mode-set = (int) [ 0, 7 ], "
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"mode-change-period = (int) [ 1, MAX ], "
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"mode-change-neighbor = (boolean) { TRUE, FALSE }, "
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"maxptime = (int) [ 20, MAX ], "
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"ptime = (int) [ 20, MAX ]"
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*/
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) 16000, " "encoding-name = (string) \"AMR-WB\", "
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/* This is the default, so the peer doesn't have to specify it
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* "encoding-params = (string) \"1\", " */
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/* NOTE that all values must be strings in orde to be able to do SDP <->
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* GstCaps mapping. */
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"octet-align = (string) \"1\";"
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/* following options are not needed for a decoder
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*
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"crc = (string) { \"0\", \"1\" }, "
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"robust-sorting = (string) \"0\", "
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"interleaving = (string) \"0\""
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"mode-set = (int) [ 0, 7 ], "
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"mode-change-period = (int) [ 1, MAX ], "
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"mode-change-neighbor = (boolean) { TRUE, FALSE }, "
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"maxptime = (int) [ 20, MAX ], "
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"ptime = (int) [ 20, MAX ]"
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*/
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)
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);
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static GstStaticPadTemplate gst_rtp_amr_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/AMR, " "channels = (int) 1," "rate = (int) 8000;"
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"audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000")
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);
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static gboolean gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_amr_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp);
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#define gst_rtp_amr_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpAMRDepay, gst_rtp_amr_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
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static void
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gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * 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_amr_depay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_amr_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP AMR depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
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"Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_amr_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_amr_depay_setcaps;
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GST_DEBUG_CATEGORY_INIT (rtpamrdepay_debug, "rtpamrdepay", 0,
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"AMR/AMR-WB RTP Depayloader");
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}
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static void
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gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay)
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{
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GstRTPBaseDepayload *depayload;
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depayload = GST_RTP_BASE_DEPAYLOAD (rtpamrdepay);
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gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
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}
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static gboolean
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gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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GstCaps *srccaps;
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GstRtpAMRDepay *rtpamrdepay;
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const gchar *params;
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const gchar *str, *type;
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gint clock_rate, need_clock_rate;
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gboolean res;
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rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
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structure = gst_caps_get_structure (caps, 0);
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/* figure out the mode first and set the clock rates */
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if ((str = gst_structure_get_string (structure, "encoding-name"))) {
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if (strcmp (str, "AMR") == 0) {
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rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB;
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need_clock_rate = 8000;
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type = "audio/AMR";
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} else if (strcmp (str, "AMR-WB") == 0) {
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rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
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need_clock_rate = 16000;
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type = "audio/AMR-WB";
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} else
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goto invalid_mode;
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} else
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goto invalid_mode;
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if (!(str = gst_structure_get_string (structure, "octet-align")))
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rtpamrdepay->octet_align = FALSE;
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else
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rtpamrdepay->octet_align = (atoi (str) == 1);
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if (!(str = gst_structure_get_string (structure, "crc")))
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rtpamrdepay->crc = FALSE;
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else
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rtpamrdepay->crc = (atoi (str) == 1);
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if (rtpamrdepay->crc) {
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/* crc mode implies octet aligned mode */
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rtpamrdepay->octet_align = TRUE;
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}
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if (!(str = gst_structure_get_string (structure, "robust-sorting")))
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rtpamrdepay->robust_sorting = FALSE;
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else
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rtpamrdepay->robust_sorting = (atoi (str) == 1);
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if (rtpamrdepay->robust_sorting) {
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/* robust_sorting mode implies octet aligned mode */
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rtpamrdepay->octet_align = TRUE;
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}
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if (!(str = gst_structure_get_string (structure, "interleaving")))
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rtpamrdepay->interleaving = FALSE;
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else
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rtpamrdepay->interleaving = (atoi (str) == 1);
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if (rtpamrdepay->interleaving) {
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/* interleaving mode implies octet aligned mode */
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rtpamrdepay->octet_align = TRUE;
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}
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if (!(params = gst_structure_get_string (structure, "encoding-params")))
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rtpamrdepay->channels = 1;
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else {
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rtpamrdepay->channels = atoi (params);
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}
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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clock_rate = need_clock_rate;
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depayload->clock_rate = clock_rate;
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/* we require 1 channel, 8000 Hz, octet aligned, no CRC,
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* no robust sorting, no interleaving for now */
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if (rtpamrdepay->channels != 1)
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return FALSE;
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if (clock_rate != need_clock_rate)
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return FALSE;
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if (rtpamrdepay->octet_align != TRUE)
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return FALSE;
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if (rtpamrdepay->robust_sorting != FALSE)
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return FALSE;
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if (rtpamrdepay->interleaving != FALSE)
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return FALSE;
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srccaps = gst_caps_new_simple (type,
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"channels", G_TYPE_INT, rtpamrdepay->channels,
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"rate", G_TYPE_INT, clock_rate, NULL);
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res = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
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return res;
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/* ERRORS */
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invalid_mode:
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{
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GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
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return FALSE;
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}
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}
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/* -1 is invalid */
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static const gint nb_frame_size[16] = {
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12, 13, 15, 17, 19, 20, 26, 31,
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5, -1, -1, -1, -1, -1, -1, 0
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};
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static const gint wb_frame_size[16] = {
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17, 23, 32, 36, 40, 46, 50, 58,
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60, 5, -1, -1, -1, -1, -1, 0
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};
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static GstBuffer *
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gst_rtp_amr_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
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GstRtpAMRDepay *rtpamrdepay;
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const gint *frame_size;
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GstBuffer *outbuf = NULL;
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gint payload_len;
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GstMapInfo map;
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rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
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/* setup frame size pointer */
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if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
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frame_size = nb_frame_size;
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else
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frame_size = wb_frame_size;
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/* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
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* no robust sorting, no interleaving data is to be depayloaded */
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{
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guint8 *payload, *p, *dp;
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gint i, num_packets, num_nonempty_packets;
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gint amr_len;
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gint ILL, ILP;
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payload_len = gst_rtp_buffer_get_payload_len (rtp);
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/* need at least 2 bytes for the header */
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if (payload_len < 2)
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goto too_small;
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payload = gst_rtp_buffer_get_payload (rtp);
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/* depay CMR. The CMR is used by the sender to request
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* a new encoding mode.
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*
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* 0 1 2 3 4 5 6 7
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* +-+-+-+-+-+-+-+-+
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* | CMR |R|R|R|R|
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* +-+-+-+-+-+-+-+-+
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*/
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/* CMR = (payload[0] & 0xf0) >> 4; */
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/* strip CMR header now, pack FT and the data for the decoder */
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payload_len -= 1;
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payload += 1;
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GST_DEBUG_OBJECT (rtpamrdepay, "payload len %d", payload_len);
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if (rtpamrdepay->interleaving) {
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ILL = (payload[0] & 0xf0) >> 4;
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ILP = (payload[0] & 0x0f);
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payload_len -= 1;
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payload += 1;
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if (ILP > ILL)
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goto wrong_interleaving;
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}
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/*
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
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* +-+-+-+-+-+-+-+-+..
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* |F| FT |Q|P|P| more FT..
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* +-+-+-+-+-+-+-+-+..
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*/
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/* count number of packets by counting the FTs. Also
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* count number of amr data bytes and number of non-empty
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* packets (this is also the number of CRCs if present). */
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amr_len = 0;
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num_nonempty_packets = 0;
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num_packets = 0;
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for (i = 0; i < payload_len; i++) {
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gint fr_size;
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guint8 FT;
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FT = (payload[i] & 0x78) >> 3;
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fr_size = frame_size[FT];
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GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
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if (fr_size == -1)
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goto wrong_framesize;
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if (fr_size > 0) {
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amr_len += fr_size;
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num_nonempty_packets++;
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}
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num_packets++;
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if ((payload[i] & 0x80) == 0)
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break;
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}
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if (rtpamrdepay->crc) {
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/* data len + CRC len + header bytes should be smaller than payload_len */
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if (num_packets + num_nonempty_packets + amr_len > payload_len)
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goto wrong_length_1;
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} else {
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/* data len + header bytes should be smaller than payload_len */
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if (num_packets + amr_len > payload_len)
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goto wrong_length_2;
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}
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outbuf = gst_buffer_new_and_alloc (payload_len);
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/* point to destination */
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gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
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/* point to first data packet */
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p = map.data;
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dp = payload + num_packets;
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if (rtpamrdepay->crc) {
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/* skip CRC if present */
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dp += num_nonempty_packets;
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}
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for (i = 0; i < num_packets; i++) {
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gint fr_size;
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/* copy FT, clear F bit */
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*p++ = payload[i] & 0x7f;
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fr_size = frame_size[(payload[i] & 0x78) >> 3];
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if (fr_size > 0) {
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/* copy data packet, FIXME, calc CRC here. */
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memcpy (p, dp, fr_size);
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p += fr_size;
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dp += fr_size;
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}
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}
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gst_buffer_unmap (outbuf, &map);
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/* we can set the duration because each packet is 20 milliseconds */
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GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
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if (gst_rtp_buffer_get_marker (rtp)) {
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/* marker bit marks a buffer after a talkspurt. */
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GST_DEBUG_OBJECT (depayload, "marker bit was set");
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
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}
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GST_DEBUG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
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gst_buffer_get_size (outbuf));
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gst_rtp_copy_audio_meta (rtpamrdepay, outbuf, rtp->buffer);
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}
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return outbuf;
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/* ERRORS */
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too_small:
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{
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GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
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(NULL), ("AMR RTP payload too small (%d)", payload_len));
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goto bad_packet;
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}
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wrong_interleaving:
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{
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GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
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(NULL), ("AMR RTP wrong interleaving"));
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goto bad_packet;
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}
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wrong_framesize:
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{
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GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
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(NULL), ("AMR RTP frame size == -1"));
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goto bad_packet;
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}
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wrong_length_1:
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{
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GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
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(NULL), ("AMR RTP wrong length 1"));
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goto bad_packet;
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}
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wrong_length_2:
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{
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GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
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(NULL), ("AMR RTP wrong length 2"));
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goto bad_packet;
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}
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bad_packet:
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{
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/* no fatal error */
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return NULL;
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}
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}
|
|
|
|
gboolean
|
|
gst_rtp_amr_depay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpamrdepay",
|
|
GST_RANK_SECONDARY, GST_TYPE_RTP_AMR_DEPAY);
|
|
}
|