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28906ed743
Original commit message from CVS: * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init), (gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_parse_audio_config), (gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps), (gst_rtp_mp4g_pay_flush): * gst/rtp/gstrtpmp4gpay.h: Make more things work. Handle ACC config strings.
570 lines
16 KiB
C
570 lines
16 KiB
C
/* GStreamer
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* Copyright (C) <2006> 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 "gstrtpmp4gpay.h"
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GST_DEBUG_CATEGORY (rtpmp4gpay_debug);
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#define GST_CAT_DEFAULT (rtpmp4gpay_debug)
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/* elementfactory information */
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static GstElementDetails gst_rtp_mp4gpay_details = {
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"RTP packet parser",
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"Codec/Payloader/Network",
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"Payload MPEG4 elementary streams as RTP packets (RFC 3640)",
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"Wim Taymans <wim@fluendo.com>"
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};
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static GstStaticPadTemplate gst_rtp_mp4g_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 ("video/mpeg,"
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"mpegversion=(int) 4,"
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"systemstream=(boolean)false;" "audio/mpeg," "mpegversion=(int) 4")
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);
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static GstStaticPadTemplate gst_rtp_mp4g_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) \"video\", "
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"payload = (int) [ 96, 127 ], "
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"clock-rate = (int) [1, MAX ], "
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"encoding-name = (string) \"mpeg4-generic\", "
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/* required string params */
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"streamtype = (string) { \"4\", \"5\" }, " /* 4 = video, 5 = audio */
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"profile-level-id = (int) [1,MAX], "
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/* "config = (string) [1,MAX]" */
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"mode = (string) { \"generic\", \"CELP-cbr\", \"CELP-vbr\", \"AAC-lbr\", \"AAC-hbr\" }, "
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/* Optional general parameters */
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"objecttype = (int) [1,MAX], " "constantsize = (int) [1,MAX], " /* constant size of each AU */
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"constantduration = (int) [1,MAX], " /* constant duration of each AU */
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"maxdisplacement = (int) [1,MAX], "
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"de-interleavebuffersize: = (int) [1,MAX], "
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/* Optional configuration parameters */
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"sizelength = (int) [1, 16], " /* max 16 bits, should be enough... */
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"indexlength = (int) [1, 8], "
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"indexdeltalength = (int) [1, 8], "
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"ctsdeltalength = (int) [1, 64], "
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"dtsdeltalength = (int) [1, 64], "
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"randomaccessindication = (int) {0, 1}, "
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"streamstateindication = (int) [0, 64], "
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"auxiliarydatasizelength = (int) [0, 64]")
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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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static void gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass);
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static void gst_rtp_mp4g_pay_base_init (GstRtpMP4GPayClass * klass);
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static void gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay);
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static void gst_rtp_mp4g_pay_finalize (GObject * object);
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static void gst_rtp_mp4g_pay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_mp4g_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buffer);
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static GstBaseRTPPayloadClass *parent_class = NULL;
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static GType
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gst_rtp_mp4g_pay_get_type (void)
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{
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static GType rtpmp4gpay_type = 0;
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if (!rtpmp4gpay_type) {
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static const GTypeInfo rtpmp4gpay_info = {
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sizeof (GstRtpMP4GPayClass),
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(GBaseInitFunc) gst_rtp_mp4g_pay_base_init,
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NULL,
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(GClassInitFunc) gst_rtp_mp4g_pay_class_init,
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NULL,
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NULL,
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sizeof (GstRtpMP4GPay),
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0,
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(GInstanceInitFunc) gst_rtp_mp4g_pay_init,
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};
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rtpmp4gpay_type =
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g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMP4GPay",
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&rtpmp4gpay_info, 0);
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}
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return rtpmp4gpay_type;
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}
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static void
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gst_rtp_mp4g_pay_base_init (GstRtpMP4GPayClass * 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_mp4g_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_mp4g_pay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_mp4gpay_details);
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}
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static void
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gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * 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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gobject_class->set_property = gst_rtp_mp4g_pay_set_property;
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gobject_class->get_property = gst_rtp_mp4g_pay_get_property;
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gobject_class->finalize = gst_rtp_mp4g_pay_finalize;
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gstbasertppayload_class->set_caps = gst_rtp_mp4g_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpmp4gpay_debug, "rtpmp4gpay", 0,
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"MP4-generic RTP Payloader");
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}
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static void
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gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay)
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{
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rtpmp4gpay->adapter = gst_adapter_new ();
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rtpmp4gpay->rate = 90000;
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rtpmp4gpay->profile = 1;
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rtpmp4gpay->mode = "";
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}
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static void
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gst_rtp_mp4g_pay_finalize (GObject * object)
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{
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GstRtpMP4GPay *rtpmp4gpay;
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rtpmp4gpay = GST_RTP_MP4G_PAY (object);
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g_object_unref (rtpmp4gpay->adapter);
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rtpmp4gpay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static unsigned sampling_table[16] = {
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96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
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16000, 12000, 11025, 8000, 7350, 0, 0, 0
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};
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static gboolean
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gst_rtp_mp4g_pay_parse_audio_config (GstRtpMP4GPay * rtpmp4gpay,
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GstBuffer * buffer)
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{
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guint8 *data;
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guint size;
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guint8 objectType;
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guint8 samplingIdx;
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guint8 channelCfg;
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data = GST_BUFFER_DATA (buffer);
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size = GST_BUFFER_SIZE (buffer);
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if (size < 2)
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goto too_short;
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/* only AAC LC for now */
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objectType = (data[0] & 0xf8) >> 3;
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if (objectType != 2)
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goto unsupported_type;
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samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
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/* only fixed values for now */
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if (samplingIdx > 12 && samplingIdx != 15)
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goto wrong_freq;
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channelCfg = ((data[1] & 0x78) >> 3);
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if (channelCfg > 7)
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goto wrong_channels;
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/* rtp rate depends on sampling rate of the audio */
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if (samplingIdx == 15) {
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if (size < 5)
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goto too_short;
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/* index of 15 means we get the rate in the next 24 bits */
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rtpmp4gpay->rate = ((data[1] & 0x7f) << 17) |
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((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7);
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} else {
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/* else use the rate from the table */
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rtpmp4gpay->rate = sampling_table[samplingIdx];
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}
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/* extra rtp params contain the number of channels */
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rtpmp4gpay->params = channelCfg;
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/* audio stream type */
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rtpmp4gpay->streamtype = 5;
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/* mode */
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rtpmp4gpay->mode = "ACC-hbr";
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/* profile (should be 1) */
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rtpmp4gpay->profile = objectType - 1;
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GST_DEBUG_OBJECT (rtpmp4gpay,
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"objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType,
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samplingIdx, rtpmp4gpay->rate, channelCfg);
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return TRUE;
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/* ERROR */
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too_short:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
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(NULL), ("config string too short"));
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return FALSE;
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}
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unsupported_type:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
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(NULL), ("unsupported object type %d", objectType));
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return FALSE;
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}
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wrong_freq:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
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(NULL), ("unsupported frequency index %d", samplingIdx));
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return FALSE;
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}
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wrong_channels:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
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(NULL), ("unsupported number of channels %d", channelCfg));
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return FALSE;
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}
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}
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#define VOS_STARTCODE 0x000001B0
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static gboolean
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gst_rtp_mp4g_pay_parse_video_config (GstRtpMP4GPay * rtpmp4gpay,
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GstBuffer * buffer)
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{
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guint8 *data;
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guint size;
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guint32 code;
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data = GST_BUFFER_DATA (buffer);
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size = GST_BUFFER_SIZE (buffer);
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if (size < 5)
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goto too_short;
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code = GST_READ_UINT32_BE (data);
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if (code == VOS_STARTCODE) {
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/* get profile */
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rtpmp4gpay->profile = data[4];
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} else {
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GST_ELEMENT_WARNING (rtpmp4gpay, STREAM, FORMAT,
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(NULL), ("profile not found in config string"));
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rtpmp4gpay->profile = 1;
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}
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/* fixed rate */
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rtpmp4gpay->rate = 90000;
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/* video stream type */
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rtpmp4gpay->streamtype = 4;
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/* no params for video */
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rtpmp4gpay->params = 0;
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/* mode */
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rtpmp4gpay->mode = "generic";
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GST_LOG_OBJECT (rtpmp4gpay, "profile %d", rtpmp4gpay->profile);
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return TRUE;
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/* ERROR */
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too_short:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
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(NULL), ("config string too short"));
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return FALSE;
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}
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}
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static void
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gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay)
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{
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gchar *config;
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GValue v = { 0 };
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#define MP4GCAPS \
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"streamtype", G_TYPE_INT, rtpmp4gpay->streamtype, \
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"profile-level-id", G_TYPE_INT, rtpmp4gpay->profile, \
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"mode", G_TYPE_STRING, rtpmp4gpay->mode, \
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"config", G_TYPE_STRING, config, \
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"sizelength", G_TYPE_INT, 13, \
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"indexlength", G_TYPE_INT, 3, \
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"indexdeltalength", G_TYPE_INT, 3, \
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NULL
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g_value_init (&v, GST_TYPE_BUFFER);
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gst_value_set_buffer (&v, rtpmp4gpay->config);
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config = gst_value_serialize (&v);
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/* hmm, silly */
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if (rtpmp4gpay->params) {
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gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
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"encoding-params", G_TYPE_INT, rtpmp4gpay->params, MP4GCAPS);
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} else {
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gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
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MP4GCAPS);
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}
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g_value_unset (&v);
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g_free (config);
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#undef MP4GCAPS
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}
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static gboolean
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gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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GstRtpMP4GPay *rtpmp4gpay;
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GstStructure *structure;
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const GValue *codec_info;
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gboolean res = TRUE;
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rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
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structure = gst_caps_get_structure (caps, 0);
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codec_info = gst_structure_get_value (structure, "codec_info");
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if (codec_info) {
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GST_LOG_OBJECT (rtpmp4gpay, "got codec_info");
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if (G_VALUE_TYPE (codec_info) == GST_TYPE_BUFFER) {
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GstBuffer *buffer;
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const gchar *name;
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buffer = gst_value_get_buffer (codec_info);
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GST_LOG_OBJECT (rtpmp4gpay, "configuring codec_info");
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name = gst_structure_get_name (structure);
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/* parse buffer */
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if (!strcmp (name, "audio/mpeg")) {
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res = gst_rtp_mp4g_pay_parse_audio_config (rtpmp4gpay, buffer);
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} else if (!strcmp (name, "video/mpeg")) {
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res = gst_rtp_mp4g_pay_parse_video_config (rtpmp4gpay, buffer);
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} else {
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res = FALSE;
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}
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if (!res)
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goto config_failed;
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/* now we can configure the buffer */
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if (rtpmp4gpay->config)
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gst_buffer_unref (rtpmp4gpay->config);
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rtpmp4gpay->config = gst_buffer_copy (buffer);
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}
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}
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gst_basertppayload_set_options (payload, "video", TRUE, "mpeg4-generic",
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rtpmp4gpay->rate);
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gst_rtp_mp4g_pay_new_caps (rtpmp4gpay);
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return res;
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/* ERRORS */
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config_failed:
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{
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GST_DEBUG_OBJECT (rtpmp4gpay, "failed to parse config");
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return FALSE;
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}
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}
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static GstFlowReturn
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gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay)
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{
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guint avail, total;
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GstBuffer *outbuf;
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GstFlowReturn ret;
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gboolean fragmented;
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guint mtu;
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fragmented = FALSE;
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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 fragment the MPEG data
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* over multiple packets. */
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total = avail = gst_adapter_available (rtpmp4gpay->adapter);
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ret = GST_FLOW_OK;
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mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpmp4gpay);
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while (avail > 0) {
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guint towrite;
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guint8 *payload;
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guint8 *data;
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guint payload_len;
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guint packet_len;
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/* this will be the total lenght of the packet */
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packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
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/* fill one MTU or all available bytes, we need 4 spare bytes for
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* the AU header. */
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towrite = MIN (packet_len, mtu - 4);
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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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GST_DEBUG_OBJECT (rtpmp4gpay,
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"avail %d, towrite %d, packet_len %d, payload_len %d", avail, towrite,
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packet_len, payload_len);
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/* create buffer to hold the payload, also make room for the 4 header bytes. */
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outbuf = gst_rtp_buffer_new_allocate (payload_len + 4, 0, 0);
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/* copy payload */
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payload = gst_rtp_buffer_get_payload (outbuf);
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/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
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* |AU-headers-length|AU-header|AU-header| |AU-header|padding|
|
|
* | | (1) | (2) | | (n) | bits |
|
|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
|
|
*/
|
|
/* AU-headers-length, we only have 1 AU-header */
|
|
payload[0] = 0x00;
|
|
payload[1] = 0x10; /* we use 16 bits for the header */
|
|
|
|
/* +---------------------------------------+
|
|
* | AU-size |
|
|
* +---------------------------------------+
|
|
* | AU-Index / AU-Index-delta |
|
|
* +---------------------------------------+
|
|
* | CTS-flag |
|
|
* +---------------------------------------+
|
|
* | CTS-delta |
|
|
* +---------------------------------------+
|
|
* | DTS-flag |
|
|
* +---------------------------------------+
|
|
* | DTS-delta |
|
|
* +---------------------------------------+
|
|
* | RAP-flag |
|
|
* +---------------------------------------+
|
|
* | Stream-state |
|
|
* +---------------------------------------+
|
|
*/
|
|
/* The AU-header, no CTS, DTS, RAP, Stream-state
|
|
*
|
|
* AU-size is always the total size of the AU, not the fragmented size
|
|
*/
|
|
payload[2] = (total & 0x1fe0) >> 5;
|
|
payload[3] = (total & 0x1f) << 3; /* we use 13 bits for the size, 3 bits index */
|
|
|
|
data = (guint8 *) gst_adapter_peek (rtpmp4gpay->adapter, payload_len);
|
|
memcpy (&payload[4], data, payload_len);
|
|
|
|
gst_adapter_flush (rtpmp4gpay->adapter, payload_len);
|
|
|
|
/* marker only if the packet is complete */
|
|
gst_rtp_buffer_set_marker (outbuf, avail > payload_len);
|
|
|
|
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_ts;
|
|
|
|
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf);
|
|
|
|
avail -= payload_len;
|
|
fragmented = TRUE;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* we expect buffers as exactly one complete AU
|
|
*/
|
|
static GstFlowReturn
|
|
gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpMP4GPay *rtpmp4gpay;
|
|
GstFlowReturn ret;
|
|
|
|
ret = GST_FLOW_OK;
|
|
|
|
rtpmp4gpay = GST_RTP_MP4G_PAY (basepayload);
|
|
|
|
rtpmp4gpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
|
|
|
|
/* we always encode and flush a full AU */
|
|
gst_adapter_push (rtpmp4gpay->adapter, buffer);
|
|
ret = gst_rtp_mp4g_pay_flush (rtpmp4gpay);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_mp4g_pay_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpMP4GPay *rtpmp4gpay;
|
|
|
|
rtpmp4gpay = GST_RTP_MP4G_PAY (object);
|
|
|
|
switch (prop_id) {
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_mp4g_pay_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpMP4GPay *rtpmp4gpay;
|
|
|
|
rtpmp4gpay = GST_RTP_MP4G_PAY (object);
|
|
|
|
switch (prop_id) {
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_mp4g_pay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpmp4gpay",
|
|
GST_RANK_NONE, GST_TYPE_RTP_MP4G_PAY);
|
|
}
|