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a297069e16
Original commit message from CVS: * gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain): Handle multiple AMr packets per payload. Handle CRC and parse ILL/ILP. * gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_setcaps): Make caps params strings for easy SDP mapping. * gst/rtp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps): Handle capsnego better. * gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_setcaps): * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_new_caps): Generate and parse config string in the caps.
225 lines
6.2 KiB
C
225 lines
6.2 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
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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 "gstrtpamrenc.h"
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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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/* elementfactory information */
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static GstElementDetails gst_rtp_amrenc_details = {
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"RTP packet parser",
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"Codec/Parser/Network",
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"Encode AMR audio into RTP packets (RFC 3267)",
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"Wim Taymans <wim@fluendo.com>"
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};
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static GstStaticPadTemplate gst_rtpamrenc_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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);
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static GstStaticPadTemplate gst_rtpamrenc_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) [ 96, 255 ], "
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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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);
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static void gst_rtpamrenc_class_init (GstRtpAMREncClass * klass);
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static void gst_rtpamrenc_base_init (GstRtpAMREncClass * klass);
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static void gst_rtpamrenc_init (GstRtpAMREnc * rtpamrenc);
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static gboolean gst_rtpamrenc_setcaps (GstBaseRTPPayload * basepayload,
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GstCaps * caps);
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static GstFlowReturn gst_rtpamrenc_handle_buffer (GstBaseRTPPayload * pad,
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GstBuffer * buffer);
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static GstBaseRTPPayloadClass *parent_class = NULL;
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static GType
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gst_rtpamrenc_get_type (void)
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{
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static GType rtpamrenc_type = 0;
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if (!rtpamrenc_type) {
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static const GTypeInfo rtpamrenc_info = {
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sizeof (GstRtpAMREncClass),
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(GBaseInitFunc) gst_rtpamrenc_base_init,
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NULL,
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(GClassInitFunc) gst_rtpamrenc_class_init,
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NULL,
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NULL,
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sizeof (GstRtpAMREnc),
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0,
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(GInstanceInitFunc) gst_rtpamrenc_init,
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};
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rtpamrenc_type =
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g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpAMREnc",
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&rtpamrenc_info, 0);
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}
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return rtpamrenc_type;
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}
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static void
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gst_rtpamrenc_base_init (GstRtpAMREncClass * 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_rtpamrenc_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtpamrenc_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_amrenc_details);
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}
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static void
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gst_rtpamrenc_class_init (GstRtpAMREncClass * 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_ref (GST_TYPE_BASE_RTP_PAYLOAD);
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gstbasertppayload_class->set_caps = gst_rtpamrenc_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtpamrenc_handle_buffer;
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}
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static void
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gst_rtpamrenc_init (GstRtpAMREnc * rtpamrenc)
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{
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}
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static gboolean
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gst_rtpamrenc_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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{
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GstRtpAMREnc *rtpamrenc;
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rtpamrenc = GST_RTP_AMR_ENC (basepayload);
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gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
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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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}
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static GstFlowReturn
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gst_rtpamrenc_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpAMREnc *rtpamrenc;
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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;
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GstClockTime timestamp;
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rtpamrenc = GST_RTP_AMR_ENC (basepayload);
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size = GST_BUFFER_SIZE (buffer);
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timestamp = GST_BUFFER_TIMESTAMP (buffer);
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/* FIXME, only one AMR frame per RTP packet for now,
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* octet aligned, no interleaving, single channel, no CRC,
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* no robust-sorting. */
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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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outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
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/* FIXME, assert for now */
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g_assert (GST_BUFFER_SIZE (outbuf) < GST_BASE_RTP_PAYLOAD_MTU (rtpamrenc));
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/* copy timestamp */
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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/* get payload */
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payload = gst_rtpbuffer_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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data = GST_BUFFER_DATA (buffer);
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/* copy data in payload */
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memcpy (&payload[1], data, 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|
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* +-+-+-+-+-+-+-+-+
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*/
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/* clear F flag */
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payload[1] = payload[1] & 0x7f;
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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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}
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gboolean
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gst_rtpamrenc_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpamrenc",
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GST_RANK_NONE, GST_TYPE_RTP_AMR_ENC);
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}
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