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Propagate the DISCONT flag to the first RTP packet being used to transfer a DISCONT buffer. https://bugzilla.gnome.org/show_bug.cgi?id=730563
641 lines
18 KiB
C
641 lines
18 KiB
C
/* GStreamer
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* Copyright (C) <2006> 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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#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/base/gstbitreader.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpmp4gpay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpmp4gpay_debug);
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#define GST_CAT_DEFAULT (rtpmp4gpay_debug)
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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;"
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"audio/mpeg," "mpegversion=(int) 4, " "stream-format=(string) raw")
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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\", \"audio\", \"application\" }, "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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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 = (string) [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 = (string) [1,MAX], " */
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/* "constantsize = (string) [1,MAX], " *//* constant size of each AU */
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/* "constantduration = (string) [1,MAX], " *//* constant duration of each AU */
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/* "maxdisplacement = (string) [1,MAX], " */
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/* "de-interleavebuffersize = (string) [1,MAX], " */
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/* Optional configuration parameters */
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/* "sizelength = (string) [1, 16], " *//* max 16 bits, should be enough... */
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/* "indexlength = (string) [1, 8], " */
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/* "indexdeltalength = (string) [1, 8], " */
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/* "ctsdeltalength = (string) [1, 64], " */
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/* "dtsdeltalength = (string) [1, 64], " */
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/* "randomaccessindication = (string) {0, 1}, " */
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/* "streamstateindication = (string) [0, 64], " */
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/* "auxiliarydatasizelength = (string) [0, 64]" */ )
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);
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static void gst_rtp_mp4g_pay_finalize (GObject * object);
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static GstStateChangeReturn gst_rtp_mp4g_pay_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_rtp_mp4g_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstRTPBasePayload *
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payload, GstBuffer * buffer);
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static gboolean gst_rtp_mp4g_pay_sink_event (GstRTPBasePayload * payload,
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GstEvent * event);
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#define gst_rtp_mp4g_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpMP4GPay, gst_rtp_mp4g_pay, GST_TYPE_RTP_BASE_PAYLOAD)
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static void 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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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gobject_class->finalize = gst_rtp_mp4g_pay_finalize;
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gstelement_class->change_state = gst_rtp_mp4g_pay_change_state;
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gstrtpbasepayload_class->set_caps = gst_rtp_mp4g_pay_setcaps;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer;
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gstrtpbasepayload_class->sink_event = gst_rtp_mp4g_pay_sink_event;
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gst_element_class_add_pad_template (gstelement_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 (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_mp4g_pay_sink_template));
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP MPEG4 ES payloader",
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"Codec/Payloader/Network/RTP",
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"Payload MPEG4 elementary streams as RTP packets (RFC 3640)",
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"Wim Taymans <wim.taymans@gmail.com>");
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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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}
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static void
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gst_rtp_mp4g_pay_reset (GstRtpMP4GPay * rtpmp4gpay)
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{
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GST_DEBUG_OBJECT (rtpmp4gpay, "reset");
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gst_adapter_clear (rtpmp4gpay->adapter);
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rtpmp4gpay->offset = 0;
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}
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static void
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gst_rtp_mp4g_pay_cleanup (GstRtpMP4GPay * rtpmp4gpay)
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{
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gst_rtp_mp4g_pay_reset (rtpmp4gpay);
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g_free (rtpmp4gpay->params);
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rtpmp4gpay->params = NULL;
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if (rtpmp4gpay->config)
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gst_buffer_unref (rtpmp4gpay->config);
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rtpmp4gpay->config = NULL;
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g_free (rtpmp4gpay->profile);
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rtpmp4gpay->profile = NULL;
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rtpmp4gpay->streamtype = NULL;
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rtpmp4gpay->mode = NULL;
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rtpmp4gpay->frame_len = 0;
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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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gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
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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 const unsigned int 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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GstMapInfo map;
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guint8 objectType = 0;
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guint8 samplingIdx = 0;
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guint8 channelCfg = 0;
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GstBitReader br;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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gst_bit_reader_init (&br, map.data, map.size);
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/* any object type is fine, we need to copy it to the profile-level-id field. */
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if (!gst_bit_reader_get_bits_uint8 (&br, &objectType, 5))
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goto too_short;
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if (objectType == 0)
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goto invalid_object;
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if (!gst_bit_reader_get_bits_uint8 (&br, &samplingIdx, 4))
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goto too_short;
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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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if (!gst_bit_reader_get_bits_uint8 (&br, &channelCfg, 4))
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goto too_short;
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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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guint32 rate = 0;
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/* index of 15 means we get the rate in the next 24 bits */
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if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24))
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goto too_short;
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rtpmp4gpay->rate = rate;
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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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rtpmp4gpay->frame_len = 1024;
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switch (objectType) {
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case 1:
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case 2:
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case 3:
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case 4:
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case 6:
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case 7:
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{
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guint8 frameLenFlag = 0;
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if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1))
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if (frameLenFlag)
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rtpmp4gpay->frame_len = 960;
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break;
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}
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default:
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break;
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}
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/* extra rtp params contain the number of channels */
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g_free (rtpmp4gpay->params);
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rtpmp4gpay->params = g_strdup_printf ("%d", channelCfg);
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/* audio stream type */
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rtpmp4gpay->streamtype = "5";
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/* mode only high bitrate for now */
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rtpmp4gpay->mode = "AAC-hbr";
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/* profile */
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g_free (rtpmp4gpay->profile);
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rtpmp4gpay->profile = g_strdup_printf ("%d", objectType);
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GST_DEBUG_OBJECT (rtpmp4gpay,
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"objectType: %d, samplingIdx: %d (%d), channelCfg: %d, frame_len %d",
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objectType, samplingIdx, rtpmp4gpay->rate, channelCfg,
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rtpmp4gpay->frame_len);
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gst_buffer_unmap (buffer, &map);
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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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gst_buffer_unmap (buffer, &map);
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return FALSE;
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}
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invalid_object:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
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(NULL), ("invalid object type"));
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gst_buffer_unmap (buffer, &map);
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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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gst_buffer_unmap (buffer, &map);
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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, must < 8", channelCfg));
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gst_buffer_unmap (buffer, &map);
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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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GstMapInfo map;
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guint32 code;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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if (map.size < 5)
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goto too_short;
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code = GST_READ_UINT32_BE (map.data);
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g_free (rtpmp4gpay->profile);
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if (code == VOS_STARTCODE) {
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/* get profile */
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rtpmp4gpay->profile = g_strdup_printf ("%d", (gint) map.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, assuming \'1\'"));
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rtpmp4gpay->profile = g_strdup ("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 = NULL;
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/* mode */
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rtpmp4gpay->mode = "generic";
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GST_LOG_OBJECT (rtpmp4gpay, "profile %s", rtpmp4gpay->profile);
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gst_buffer_unmap (buffer, &map);
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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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gst_buffer_unmap (buffer, &map);
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return FALSE;
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}
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}
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static gboolean
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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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gboolean res;
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#define MP4GCAPS \
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"streamtype", G_TYPE_STRING, rtpmp4gpay->streamtype, \
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"profile-level-id", G_TYPE_STRING, 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_STRING, "13", \
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"indexlength", G_TYPE_STRING, "3", \
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"indexdeltalength", G_TYPE_STRING, "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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res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4gpay),
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"encoding-params", G_TYPE_STRING, rtpmp4gpay->params, MP4GCAPS);
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} else {
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res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_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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return res;
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}
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static gboolean
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gst_rtp_mp4g_pay_setcaps (GstRTPBasePayload * 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_data;
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const gchar *media_type = NULL;
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gboolean res;
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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_data = gst_structure_get_value (structure, "codec_data");
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if (codec_data) {
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GST_LOG_OBJECT (rtpmp4gpay, "got codec_data");
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if (G_VALUE_TYPE (codec_data) == 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_data);
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GST_LOG_OBJECT (rtpmp4gpay, "configuring codec_data");
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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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media_type = "audio";
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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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media_type = "video";
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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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if (media_type == NULL)
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goto config_failed;
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gst_rtp_base_payload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC",
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rtpmp4gpay->rate);
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res = 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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guint mtu;
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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_RTP_BASE_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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guint payload_len;
|
|
guint packet_len;
|
|
GstRTPBuffer rtp = { NULL };
|
|
GstBuffer *paybuf;
|
|
|
|
/* this will be the total lenght of the packet */
|
|
packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
|
|
|
|
/* fill one MTU or all available bytes, we need 4 spare bytes for
|
|
* the AU header. */
|
|
towrite = MIN (packet_len, mtu - 4);
|
|
|
|
/* this is the payload length */
|
|
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
|
|
|
|
GST_DEBUG_OBJECT (rtpmp4gpay,
|
|
"avail %d, towrite %d, packet_len %d, payload_len %d", avail, towrite,
|
|
packet_len, payload_len);
|
|
|
|
/* create buffer to hold the payload, also make room for the 4 header bytes. */
|
|
outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);
|
|
|
|
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
|
|
|
|
/* copy payload */
|
|
payload = gst_rtp_buffer_get_payload (&rtp);
|
|
|
|
/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
|
|
* |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 */
|
|
|
|
/* marker only if the packet is complete */
|
|
gst_rtp_buffer_set_marker (&rtp, avail <= payload_len);
|
|
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
paybuf = gst_adapter_take_buffer_fast (rtpmp4gpay->adapter, payload_len);
|
|
outbuf = gst_buffer_append (outbuf, paybuf);
|
|
|
|
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_timestamp;
|
|
GST_BUFFER_DURATION (outbuf) = rtpmp4gpay->first_duration;
|
|
|
|
if (rtpmp4gpay->frame_len) {
|
|
GST_BUFFER_OFFSET (outbuf) = rtpmp4gpay->offset;
|
|
rtpmp4gpay->offset += rtpmp4gpay->frame_len;
|
|
}
|
|
|
|
if (rtpmp4gpay->discont) {
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
|
|
/* Only the first outputted buffer has the DISCONT flag */
|
|
rtpmp4gpay->discont = FALSE;
|
|
}
|
|
|
|
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp4gpay), outbuf);
|
|
|
|
avail -= payload_len;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* we expect buffers as exactly one complete AU
|
|
*/
|
|
static GstFlowReturn
|
|
gst_rtp_mp4g_pay_handle_buffer (GstRTPBasePayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpMP4GPay *rtpmp4gpay;
|
|
|
|
rtpmp4gpay = GST_RTP_MP4G_PAY (basepayload);
|
|
|
|
rtpmp4gpay->first_timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
rtpmp4gpay->first_duration = GST_BUFFER_DURATION (buffer);
|
|
rtpmp4gpay->discont = GST_BUFFER_IS_DISCONT (buffer);
|
|
|
|
/* we always encode and flush a full AU */
|
|
gst_adapter_push (rtpmp4gpay->adapter, buffer);
|
|
|
|
return gst_rtp_mp4g_pay_flush (rtpmp4gpay);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_mp4g_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
|
|
{
|
|
GstRtpMP4GPay *rtpmp4gpay;
|
|
|
|
rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
|
|
|
|
GST_DEBUG ("Got event: %s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEGMENT:
|
|
case GST_EVENT_EOS:
|
|
/* This flush call makes sure that the last buffer is always pushed
|
|
* to the base payloader */
|
|
gst_rtp_mp4g_pay_flush (rtpmp4gpay);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_rtp_mp4g_pay_reset (rtpmp4gpay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* let parent handle event too */
|
|
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_mp4g_pay_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret;
|
|
GstRtpMP4GPay *rtpmp4gpay;
|
|
|
|
rtpmp4gpay = GST_RTP_MP4G_PAY (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_mp4g_pay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpmp4gpay",
|
|
GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_PAY);
|
|
}
|