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e7f919986a
Original commit message from CVS: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: Remove copy/paste unused code (property setters and getter) found by the coverage suite (yay, saves ~20k on disk).
368 lines
11 KiB
C
368 lines
11 KiB
C
/* GStreamer
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* Copyright (C) <2005> Wim Taymans <wim@fluendo.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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpamrpay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug);
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#define GST_CAT_DEFAULT (rtpamrpay_debug)
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/* references:
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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
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* Multi-Rate Wideband (AMR-WB) Audio Codecs.
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*
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* ETSI TS 126 201 V6.0.0 (2004-12) - Digital cellular telecommunications system (Phase 2+);
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* Universal Mobile Telecommunications System (UMTS);
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* AMR speech codec, wideband;
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* Frame structure
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* (3GPP TS 26.201 version 6.0.0 Release 6)
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*/
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/* elementfactory information */
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static const GstElementDetails gst_rtp_amrpay_details =
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GST_ELEMENT_DETAILS ("RTP packet payloader",
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"Codec/Payloader/Network",
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"Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267)",
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"Wim Taymans <wim@fluendo.com>");
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static GstStaticPadTemplate gst_rtp_amr_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/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 GstStaticPadTemplate gst_rtp_amr_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) 8000, "
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"encoding-name = (string) \"AMR\", "
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"encoding-params = (string) \"1\", "
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"octet-align = (string) \"1\", "
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"crc = (string) \"0\", "
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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 = (string) { \"0\", \"1\" }, "
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"maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ];"
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 16000, "
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"encoding-name = (string) \"AMR-WB\", "
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"encoding-params = (string) \"1\", "
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"octet-align = (string) \"1\", "
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"crc = (string) \"0\", "
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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 = (string) { \"0\", \"1\" }, "
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"maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ]")
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);
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static gboolean gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * pad,
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GstBuffer * buffer);
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GST_BOILERPLATE (GstRtpAMRPay, gst_rtp_amr_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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static void
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gst_rtp_amr_pay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_amr_pay_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_amr_pay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_amrpay_details);
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}
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static void
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gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gstbasertppayload_class->set_caps = gst_rtp_amr_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_amr_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpamrpay_debug, "rtpamrpay", 0,
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"AMR/AMR-WB RTP Payloader");
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}
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static void
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gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay, GstRtpAMRPayClass * klass)
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{
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/* needed because of GST_BOILERPLATE */
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}
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static gboolean
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gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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{
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GstRtpAMRPay *rtpamrpay;
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const GstStructure *s;
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const gchar *str;
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rtpamrpay = GST_RTP_AMR_PAY (basepayload);
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/* figure out the mode Narrow or Wideband */
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s = gst_caps_get_structure (caps, 0);
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if ((str = gst_structure_get_name (s))) {
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if (strcmp (str, "audio/AMR") == 0)
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rtpamrpay->mode = GST_RTP_AMR_P_MODE_NB;
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else if (strcmp (str, "audio/AMR-WB") == 0)
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rtpamrpay->mode = GST_RTP_AMR_P_MODE_WB;
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else
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goto wrong_type;
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} else
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goto wrong_type;
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if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
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gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
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else
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gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR-WB",
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16000);
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gst_basertppayload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1",
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/* don't set the defaults
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*
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* "crc", G_TYPE_STRING, "0",
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* "robust-sorting", G_TYPE_STRING, "0",
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* "interleaving", G_TYPE_STRING, "0",
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*/
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NULL);
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return TRUE;
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/* ERRORS */
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wrong_type:
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{
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GST_ERROR_OBJECT (rtpamrpay, "unsupported media type '%s'",
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GST_STR_NULL (str));
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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 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 gint wb_frame_size[16] = {
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17, 23, 32, 36, 40, 46, 50, 58,
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60, -1, -1, -1, -1, -1, -1, 0
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};
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static GstFlowReturn
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gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpAMRPay *rtpamrpay;
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GstFlowReturn ret;
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guint size, payload_len;
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GstBuffer *outbuf;
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guint8 *payload, *data, *payload_amr;
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GstClockTime timestamp, duration;
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guint packet_len, mtu;
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gint i, num_packets, num_nonempty_packets;
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gint amr_len;
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gint *frame_size;
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rtpamrpay = GST_RTP_AMR_PAY (basepayload);
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mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpamrpay);
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size = GST_BUFFER_SIZE (buffer);
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data = GST_BUFFER_DATA (buffer);
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timestamp = GST_BUFFER_TIMESTAMP (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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/* setup frame size pointer */
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if (rtpamrpay->mode == GST_RTP_AMR_P_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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GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
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/* FIXME, only
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* octet aligned, no interleaving, single channel, no CRC,
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* no robust-sorting. To fix this you need to implement the downstream
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* negotiation function. */
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/* first count number of packets and total amr frame size */
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amr_len = num_packets = num_nonempty_packets = 0;
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for (i = 0; i < size; i++) {
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guint8 FT;
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gint fr_size;
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FT = (data[i] & 0x78) >> 3;
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fr_size = frame_size[FT];
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GST_DEBUG_OBJECT (basepayload, "frame size %d", fr_size);
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/* FIXME, we don't handle this yet.. */
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if (fr_size <= 0)
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goto wrong_size;
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amr_len += fr_size;
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num_nonempty_packets++;
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num_packets++;
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i += fr_size;
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}
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if (amr_len > size)
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goto incomplete_frame;
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/* we need one extra byte for the CMR, the ToC is in the input
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* data */
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payload_len = size + 1;
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/* get packet len to check against MTU */
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packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
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if (packet_len > mtu)
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goto too_big;
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/* now alloc output buffer */
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outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
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/* copy timestamp, or fabricate one */
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if (timestamp != GST_CLOCK_TIME_NONE)
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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else {
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/* AMR (nb) and AMR-WB both have 20 ms per frame */
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/* FIXME: when we do more than one AMR frame per packet, fix this */
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gint count = basepayload->seqnum - basepayload->seqnum_base;
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GST_BUFFER_TIMESTAMP (outbuf) = count * 20 * GST_MSECOND;
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}
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if (duration != GST_CLOCK_TIME_NONE)
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GST_BUFFER_DURATION (outbuf) = duration;
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else {
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GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
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}
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/* get payload, this is now writable */
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payload = gst_rtp_buffer_get_payload (outbuf);
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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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payload[0] = 0xF0; /* CMR, no specific mode requested */
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/* this is where we copy the AMR data, after num_packets FTs and the
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* CMR. */
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payload_amr = payload + num_packets + 1;
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/* copy data in payload, first we copy all the FTs then all
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* the AMR data. The last FT has to have the F flag cleared. */
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for (i = 1; i <= num_packets; i++) {
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guint8 FT;
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gint fr_size;
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/* 0 1 2 3 4 5 6 7
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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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FT = (*data & 0x78) >> 3;
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fr_size = frame_size[FT];
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if (i == num_packets)
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/* last packet, clear F flag */
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payload[i] = *data & 0x7f;
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else
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/* set F flag */
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payload[i] = *data | 0x80;
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memcpy (payload_amr, &data[1], fr_size);
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/* all sizes are > 0 since we checked for that above */
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data += fr_size + 1;
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payload_amr += fr_size;
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}
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gst_buffer_unref (buffer);
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ret = gst_basertppayload_push (basepayload, outbuf);
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return ret;
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/* ERRORS */
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wrong_size:
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{
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GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
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(NULL), ("received AMR frame with size <= 0"));
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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incomplete_frame:
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{
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GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
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(NULL), ("received incomplete AMR frames"));
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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too_big:
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{
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GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
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(NULL), ("received too many AMR frames for MTU"));
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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}
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gboolean
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gst_rtp_amr_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpamrpay",
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GST_RANK_NONE, GST_TYPE_RTP_AMR_PAY);
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}
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