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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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964 lines
29 KiB
C
964 lines
29 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/rtp/gstrtpbuffer.h>
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#include <gst/audio/audio.h>
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#include "fnv1hash.h"
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#include "gstrtpvorbispay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpvorbispay_debug);
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#define GST_CAT_DEFAULT (rtpvorbispay_debug)
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/* references:
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* http://www.rfc-editor.org/rfc/rfc5215.txt
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*/
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static GstStaticPadTemplate gst_rtp_vorbis_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) [1, MAX ], " "encoding-name = (string) \"VORBIS\""
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/* All required parameters
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*
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* "encoding-params = (string) <num channels>"
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* "configuration = (string) ANY"
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*/
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)
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);
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static GstStaticPadTemplate gst_rtp_vorbis_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/x-vorbis")
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);
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#define DEFAULT_CONFIG_INTERVAL 0
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enum
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{
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PROP_0,
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PROP_CONFIG_INTERVAL
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};
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#define gst_rtp_vorbis_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpVorbisPay, gst_rtp_vorbis_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static gboolean gst_rtp_vorbis_pay_setcaps (GstRTPBasePayload * basepayload,
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GstCaps * caps);
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static GstStateChangeReturn gst_rtp_vorbis_pay_change_state (GstElement *
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element, GstStateChange transition);
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static GstFlowReturn gst_rtp_vorbis_pay_handle_buffer (GstRTPBasePayload * pad,
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GstBuffer * buffer);
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static gboolean gst_rtp_vorbis_pay_sink_event (GstRTPBasePayload * payload,
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GstEvent * event);
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static gboolean gst_rtp_vorbis_pay_parse_id (GstRTPBasePayload * basepayload,
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guint8 * data, guint size);
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static gboolean gst_rtp_vorbis_pay_finish_headers (GstRTPBasePayload *
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basepayload);
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static void gst_rtp_vorbis_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_vorbis_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void
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gst_rtp_vorbis_pay_class_init (GstRtpVorbisPayClass * 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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gstelement_class->change_state = gst_rtp_vorbis_pay_change_state;
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gstrtpbasepayload_class->set_caps = gst_rtp_vorbis_pay_setcaps;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_vorbis_pay_handle_buffer;
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gstrtpbasepayload_class->sink_event = gst_rtp_vorbis_pay_sink_event;
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gobject_class->set_property = gst_rtp_vorbis_pay_set_property;
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gobject_class->get_property = gst_rtp_vorbis_pay_get_property;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_vorbis_pay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_vorbis_pay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP Vorbis payloader",
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"Codec/Payloader/Network/RTP",
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"Payload-encode Vorbis audio into RTP packets (RFC 5215)",
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"Wim Taymans <wim.taymans@gmail.com>");
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GST_DEBUG_CATEGORY_INIT (rtpvorbispay_debug, "rtpvorbispay", 0,
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"Vorbis RTP Payloader");
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CONFIG_INTERVAL,
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g_param_spec_uint ("config-interval", "Config Send Interval",
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"Send Config Insertion Interval in seconds (configuration headers "
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"will be multiplexed in the data stream when detected.) (0 = disabled)",
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0, 3600, DEFAULT_CONFIG_INTERVAL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
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);
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}
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static void
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gst_rtp_vorbis_pay_init (GstRtpVorbisPay * rtpvorbispay)
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{
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rtpvorbispay->last_config = GST_CLOCK_TIME_NONE;
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}
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static void
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gst_rtp_vorbis_pay_clear_packet (GstRtpVorbisPay * rtpvorbispay)
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{
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if (rtpvorbispay->packet)
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gst_buffer_unref (rtpvorbispay->packet);
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rtpvorbispay->packet = NULL;
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g_list_free_full (rtpvorbispay->packet_buffers,
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(GDestroyNotify) gst_buffer_unref);
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rtpvorbispay->packet_buffers = NULL;
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}
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static void
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gst_rtp_vorbis_pay_cleanup (GstRtpVorbisPay * rtpvorbispay)
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{
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gst_rtp_vorbis_pay_clear_packet (rtpvorbispay);
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g_list_free_full (rtpvorbispay->headers, (GDestroyNotify) gst_buffer_unref);
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rtpvorbispay->headers = NULL;
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g_free (rtpvorbispay->config_data);
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rtpvorbispay->config_data = NULL;
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rtpvorbispay->last_config = GST_CLOCK_TIME_NONE;
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}
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static gboolean
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gst_rtp_vorbis_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
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{
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GstRtpVorbisPay *rtpvorbispay;
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GstStructure *s;
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const GValue *array;
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gint asize, i;
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GstBuffer *buf;
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GstMapInfo map;
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rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
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s = gst_caps_get_structure (caps, 0);
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rtpvorbispay->need_headers = TRUE;
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if ((array = gst_structure_get_value (s, "streamheader")) == NULL)
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goto done;
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if (G_VALUE_TYPE (array) != GST_TYPE_ARRAY)
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goto done;
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if ((asize = gst_value_array_get_size (array)) < 3)
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goto done;
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for (i = 0; i < asize; i++) {
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const GValue *value;
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value = gst_value_array_get_value (array, i);
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if ((buf = gst_value_get_buffer (value)) == NULL)
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goto null_buffer;
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gst_buffer_map (buf, &map, GST_MAP_READ);
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if (map.size < 1)
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goto invalid_streamheader;
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/* no data packets allowed */
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if ((map.data[0] & 1) == 0)
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goto invalid_streamheader;
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/* we need packets with id 1, 3, 5 */
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if (map.data[0] != (i * 2) + 1)
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goto invalid_streamheader;
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if (i == 0) {
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/* identification, we need to parse this in order to get the clock rate. */
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if (G_UNLIKELY (!gst_rtp_vorbis_pay_parse_id (basepayload, map.data,
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map.size)))
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goto parse_id_failed;
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}
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GST_DEBUG_OBJECT (rtpvorbispay, "collecting header %d", i);
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rtpvorbispay->headers =
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g_list_append (rtpvorbispay->headers, gst_buffer_ref (buf));
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gst_buffer_unmap (buf, &map);
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}
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if (!gst_rtp_vorbis_pay_finish_headers (basepayload))
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goto finish_failed;
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done:
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return TRUE;
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/* ERRORS */
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null_buffer:
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{
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GST_WARNING_OBJECT (rtpvorbispay, "streamheader with null buffer received");
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return FALSE;
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}
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invalid_streamheader:
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{
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GST_WARNING_OBJECT (rtpvorbispay, "unable to parse initial header");
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gst_buffer_unmap (buf, &map);
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return FALSE;
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}
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parse_id_failed:
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{
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GST_WARNING_OBJECT (rtpvorbispay, "unable to parse initial header");
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gst_buffer_unmap (buf, &map);
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return FALSE;
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}
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finish_failed:
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{
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GST_WARNING_OBJECT (rtpvorbispay, "unable to finish headers");
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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_vorbis_pay_reset_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT)
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{
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guint payload_len;
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GstRTPBuffer rtp = { NULL };
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GST_LOG_OBJECT (rtpvorbispay, "reset packet");
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rtpvorbispay->payload_pos = 4;
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gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_READ, &rtp);
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payload_len = gst_rtp_buffer_get_payload_len (&rtp);
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gst_rtp_buffer_unmap (&rtp);
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rtpvorbispay->payload_left = payload_len - 4;
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rtpvorbispay->payload_duration = 0;
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rtpvorbispay->payload_F = 0;
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rtpvorbispay->payload_VDT = VDT;
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rtpvorbispay->payload_pkts = 0;
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}
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static void
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gst_rtp_vorbis_pay_init_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT,
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GstClockTime timestamp)
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{
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GST_LOG_OBJECT (rtpvorbispay, "starting new packet, VDT: %d", VDT);
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gst_rtp_vorbis_pay_clear_packet (rtpvorbispay);
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/* new packet allocate max packet size */
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rtpvorbispay->packet =
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gst_rtp_buffer_new_allocate_len (GST_RTP_BASE_PAYLOAD_MTU
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(rtpvorbispay), 0, 0);
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gst_rtp_vorbis_pay_reset_packet (rtpvorbispay, VDT);
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GST_BUFFER_PTS (rtpvorbispay->packet) = timestamp;
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}
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static GstFlowReturn
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gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay)
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{
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GstFlowReturn ret;
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guint8 *payload;
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guint hlen;
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GstRTPBuffer rtp = { NULL };
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GList *l;
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/* check for empty packet */
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if (!rtpvorbispay->packet || rtpvorbispay->payload_pos <= 4)
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return GST_FLOW_OK;
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GST_LOG_OBJECT (rtpvorbispay, "flushing packet");
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gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_WRITE, &rtp);
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/* fix header */
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payload = gst_rtp_buffer_get_payload (&rtp);
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/*
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* 0 1 2 3
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Ident | F |VDT|# pkts.|
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*
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* F: Fragment type (0=none, 1=start, 2=cont, 3=end)
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* VDT: Vorbis data type (0=vorbis, 1=config, 2=comment, 3=reserved)
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* pkts: number of packets.
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*/
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payload[0] = (rtpvorbispay->payload_ident >> 16) & 0xff;
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payload[1] = (rtpvorbispay->payload_ident >> 8) & 0xff;
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payload[2] = (rtpvorbispay->payload_ident) & 0xff;
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payload[3] = (rtpvorbispay->payload_F & 0x3) << 6 |
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(rtpvorbispay->payload_VDT & 0x3) << 4 |
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(rtpvorbispay->payload_pkts & 0xf);
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gst_rtp_buffer_unmap (&rtp);
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/* shrink the buffer size to the last written byte */
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hlen = gst_rtp_buffer_calc_header_len (0);
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gst_buffer_resize (rtpvorbispay->packet, 0, hlen + rtpvorbispay->payload_pos);
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GST_BUFFER_DURATION (rtpvorbispay->packet) = rtpvorbispay->payload_duration;
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for (l = g_list_last (rtpvorbispay->packet_buffers); l; l = l->prev) {
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GstBuffer *buf = GST_BUFFER_CAST (l->data);
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gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpvorbispay), rtpvorbispay->packet,
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buf, g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
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gst_buffer_unref (buf);
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}
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g_list_free (rtpvorbispay->packet_buffers);
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rtpvorbispay->packet_buffers = NULL;
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/* push, this gives away our ref to the packet, so clear it. */
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ret =
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gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpvorbispay),
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rtpvorbispay->packet);
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rtpvorbispay->packet = NULL;
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return ret;
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}
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static gboolean
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gst_rtp_vorbis_pay_finish_headers (GstRTPBasePayload * basepayload)
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{
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GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
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GList *walk;
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guint length, size, n_headers, configlen, extralen;
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gchar *cstr, *configuration;
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guint8 *data, *config;
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guint32 ident;
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gboolean res;
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GST_DEBUG_OBJECT (rtpvorbispay, "finish headers");
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if (!rtpvorbispay->headers)
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goto no_headers;
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/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Number of packed headers |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Packed header |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Packed header |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | .... |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*
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* We only construct a config containing 1 packed header like this:
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*
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* 0 1 2 3
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
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* | Ident | length ..
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .. | n. of headers | length1 | length2 ..
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .. | Identification Header ..
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .................................................................
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .. | Comment Header ..
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
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* .................................................................
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .. Comment Header |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Setup Header ..
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
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* .................................................................
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .. Setup Header |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*/
|
|
|
|
/* we need 4 bytes for the number of headers (which is always 1), 3 bytes for
|
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* the ident, 2 bytes for length, 1 byte for n. of headers. */
|
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size = 4 + 3 + 2 + 1;
|
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|
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/* count the size of the headers first and update the hash */
|
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length = 0;
|
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n_headers = 0;
|
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ident = fnv1_hash_32_new ();
|
|
extralen = 1;
|
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for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) {
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GstBuffer *buf = GST_BUFFER_CAST (walk->data);
|
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GstMapInfo map;
|
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guint bsize;
|
|
|
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bsize = gst_buffer_get_size (buf);
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length += bsize;
|
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n_headers++;
|
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|
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/* count number of bytes needed for length fields, we don't need this for
|
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* the last header. */
|
|
if (g_list_next (walk)) {
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do {
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size++;
|
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extralen++;
|
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bsize >>= 7;
|
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} while (bsize);
|
|
}
|
|
/* update hash */
|
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gst_buffer_map (buf, &map, GST_MAP_READ);
|
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ident = fnv1_hash_32_update (ident, map.data, map.size);
|
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gst_buffer_unmap (buf, &map);
|
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}
|
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|
|
/* packet length is header size + packet length */
|
|
configlen = size + length;
|
|
config = data = g_malloc (configlen);
|
|
|
|
/* number of packed headers, we only pack 1 header */
|
|
data[0] = 0;
|
|
data[1] = 0;
|
|
data[2] = 0;
|
|
data[3] = 1;
|
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|
|
ident = fnv1_hash_32_to_24 (ident);
|
|
rtpvorbispay->payload_ident = ident;
|
|
GST_DEBUG_OBJECT (rtpvorbispay, "ident 0x%08x", ident);
|
|
|
|
/* take lower 3 bytes */
|
|
data[4] = (ident >> 16) & 0xff;
|
|
data[5] = (ident >> 8) & 0xff;
|
|
data[6] = ident & 0xff;
|
|
|
|
/* store length of all vorbis headers */
|
|
data[7] = ((length) >> 8) & 0xff;
|
|
data[8] = (length) & 0xff;
|
|
|
|
/* store number of headers minus one. */
|
|
data[9] = n_headers - 1;
|
|
data += 10;
|
|
|
|
/* store length for each header */
|
|
for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) {
|
|
GstBuffer *buf = GST_BUFFER_CAST (walk->data);
|
|
guint bsize, size, temp;
|
|
guint flag;
|
|
|
|
/* only need to store the length when it's not the last header */
|
|
if (!g_list_next (walk))
|
|
break;
|
|
|
|
bsize = gst_buffer_get_size (buf);
|
|
|
|
/* calc size */
|
|
size = 0;
|
|
do {
|
|
size++;
|
|
bsize >>= 7;
|
|
} while (bsize);
|
|
temp = size;
|
|
|
|
bsize = gst_buffer_get_size (buf);
|
|
/* write the size backwards */
|
|
flag = 0;
|
|
while (size) {
|
|
size--;
|
|
data[size] = (bsize & 0x7f) | flag;
|
|
bsize >>= 7;
|
|
flag = 0x80; /* Flag bit on all bytes of the length except the last */
|
|
}
|
|
data += temp;
|
|
}
|
|
|
|
/* copy header data */
|
|
for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) {
|
|
GstBuffer *buf = GST_BUFFER_CAST (walk->data);
|
|
|
|
gst_buffer_extract (buf, 0, data, gst_buffer_get_size (buf));
|
|
data += gst_buffer_get_size (buf);
|
|
}
|
|
rtpvorbispay->need_headers = FALSE;
|
|
|
|
/* serialize to base64 */
|
|
configuration = g_base64_encode (config, configlen);
|
|
|
|
/* store for later re-sending */
|
|
g_free (rtpvorbispay->config_data);
|
|
rtpvorbispay->config_size = configlen - 4 - 3 - 2;
|
|
rtpvorbispay->config_data = g_malloc (rtpvorbispay->config_size);
|
|
rtpvorbispay->config_extra_len = extralen;
|
|
memcpy (rtpvorbispay->config_data, config + 4 + 3 + 2,
|
|
rtpvorbispay->config_size);
|
|
|
|
g_free (config);
|
|
|
|
/* configure payloader settings */
|
|
cstr = g_strdup_printf ("%d", rtpvorbispay->channels);
|
|
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "VORBIS",
|
|
rtpvorbispay->rate);
|
|
res =
|
|
gst_rtp_base_payload_set_outcaps (basepayload, "encoding-params",
|
|
G_TYPE_STRING, cstr, "configuration", G_TYPE_STRING, configuration, NULL);
|
|
g_free (cstr);
|
|
g_free (configuration);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
no_headers:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpvorbispay, "finish headers");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_vorbis_pay_parse_id (GstRTPBasePayload * basepayload, guint8 * data,
|
|
guint size)
|
|
{
|
|
GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
|
|
guint8 channels;
|
|
gint32 rate, version;
|
|
|
|
if (G_UNLIKELY (size < 16))
|
|
goto too_short;
|
|
|
|
if (G_UNLIKELY (memcmp (data, "\001vorbis", 7)))
|
|
goto invalid_start;
|
|
data += 7;
|
|
|
|
if (G_UNLIKELY ((version = GST_READ_UINT32_LE (data)) != 0))
|
|
goto invalid_version;
|
|
data += 4;
|
|
|
|
if (G_UNLIKELY ((channels = *data++) < 1))
|
|
goto invalid_channels;
|
|
|
|
if (G_UNLIKELY ((rate = GST_READ_UINT32_LE (data)) < 1))
|
|
goto invalid_rate;
|
|
|
|
/* all fine, store the values */
|
|
rtpvorbispay->channels = channels;
|
|
rtpvorbispay->rate = rate;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
too_short:
|
|
{
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
|
|
("Identification packet is too short, need at least 16, got %d", size),
|
|
(NULL));
|
|
return FALSE;
|
|
}
|
|
invalid_start:
|
|
{
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
|
|
("Invalid header start in identification packet"), (NULL));
|
|
return FALSE;
|
|
}
|
|
invalid_version:
|
|
{
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
|
|
("Invalid version, expected 0, got %d", version), (NULL));
|
|
return FALSE;
|
|
}
|
|
invalid_rate:
|
|
{
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
|
|
("Invalid rate %d", rate), (NULL));
|
|
return FALSE;
|
|
}
|
|
invalid_channels:
|
|
{
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
|
|
("Invalid channels %d", channels), (NULL));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_vorbis_pay_payload_buffer (GstRtpVorbisPay * rtpvorbispay, guint8 VDT,
|
|
GstBuffer * buffer, guint8 * data, guint size, GstClockTime timestamp,
|
|
GstClockTime duration, guint not_in_length)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
guint newsize;
|
|
guint packet_len;
|
|
GstClockTime newduration;
|
|
gboolean flush;
|
|
guint plen;
|
|
guint8 *ppos, *payload;
|
|
gboolean fragmented;
|
|
GstRTPBuffer rtp = { NULL };
|
|
|
|
/* size increases with packet length and 2 bytes size eader. */
|
|
newduration = rtpvorbispay->payload_duration;
|
|
if (duration != GST_CLOCK_TIME_NONE)
|
|
newduration += duration;
|
|
|
|
newsize = rtpvorbispay->payload_pos + 2 + size;
|
|
packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0);
|
|
|
|
/* check buffer filled against length and max latency */
|
|
flush = gst_rtp_base_payload_is_filled (GST_RTP_BASE_PAYLOAD (rtpvorbispay),
|
|
packet_len, newduration);
|
|
/* we can store up to 15 vorbis packets in one RTP packet. */
|
|
flush |= (rtpvorbispay->payload_pkts == 15);
|
|
/* flush if we have a new VDT */
|
|
if (rtpvorbispay->packet)
|
|
flush |= (rtpvorbispay->payload_VDT != VDT);
|
|
if (flush)
|
|
ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
|
|
|
|
if (ret != GST_FLOW_OK)
|
|
goto done;
|
|
|
|
/* create new packet if we must */
|
|
if (!rtpvorbispay->packet) {
|
|
gst_rtp_vorbis_pay_init_packet (rtpvorbispay, VDT, timestamp);
|
|
}
|
|
|
|
gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_WRITE, &rtp);
|
|
payload = gst_rtp_buffer_get_payload (&rtp);
|
|
ppos = payload + rtpvorbispay->payload_pos;
|
|
fragmented = FALSE;
|
|
|
|
/* put buffer in packet, it either fits completely or needs to be fragmented
|
|
* over multiple RTP packets. */
|
|
do {
|
|
plen = MIN (rtpvorbispay->payload_left - 2, size);
|
|
|
|
GST_LOG_OBJECT (rtpvorbispay, "append %u bytes", plen);
|
|
|
|
/* data is copied in the payload with a 2 byte length header */
|
|
ppos[0] = ((plen - not_in_length) >> 8) & 0xff;
|
|
ppos[1] = ((plen - not_in_length) & 0xff);
|
|
if (plen)
|
|
memcpy (&ppos[2], data, plen);
|
|
|
|
if (buffer) {
|
|
if (!rtpvorbispay->packet_buffers
|
|
|| rtpvorbispay->packet_buffers->data != (gpointer) buffer)
|
|
rtpvorbispay->packet_buffers =
|
|
g_list_prepend (rtpvorbispay->packet_buffers,
|
|
gst_buffer_ref (buffer));
|
|
} else {
|
|
GList *l;
|
|
|
|
for (l = rtpvorbispay->headers; l; l = l->next)
|
|
rtpvorbispay->packet_buffers =
|
|
g_list_prepend (rtpvorbispay->packet_buffers,
|
|
gst_buffer_ref (l->data));
|
|
}
|
|
|
|
/* only first (only) configuration cuts length field */
|
|
/* NOTE: spec (if any) is not clear on this ... */
|
|
not_in_length = 0;
|
|
|
|
size -= plen;
|
|
data += plen;
|
|
|
|
rtpvorbispay->payload_pos += plen + 2;
|
|
rtpvorbispay->payload_left -= plen + 2;
|
|
|
|
if (fragmented) {
|
|
if (size == 0)
|
|
/* last fragment, set F to 0x3. */
|
|
rtpvorbispay->payload_F = 0x3;
|
|
else
|
|
/* fragment continues, set F to 0x2. */
|
|
rtpvorbispay->payload_F = 0x2;
|
|
} else {
|
|
if (size > 0) {
|
|
/* fragmented packet starts, set F to 0x1, mark ourselves as
|
|
* fragmented. */
|
|
rtpvorbispay->payload_F = 0x1;
|
|
fragmented = TRUE;
|
|
}
|
|
}
|
|
if (fragmented) {
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
/* fragmented packets are always flushed and have ptks of 0 */
|
|
rtpvorbispay->payload_pkts = 0;
|
|
ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
|
|
|
|
if (size > 0) {
|
|
/* start new packet and get pointers. VDT stays the same. */
|
|
gst_rtp_vorbis_pay_init_packet (rtpvorbispay,
|
|
rtpvorbispay->payload_VDT, timestamp);
|
|
gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_WRITE, &rtp);
|
|
payload = gst_rtp_buffer_get_payload (&rtp);
|
|
ppos = payload + rtpvorbispay->payload_pos;
|
|
}
|
|
} else {
|
|
/* unfragmented packet, update stats for next packet, size == 0 and we
|
|
* exit the while loop */
|
|
rtpvorbispay->payload_pkts++;
|
|
if (duration != GST_CLOCK_TIME_NONE)
|
|
rtpvorbispay->payload_duration += duration;
|
|
}
|
|
} while (size && ret == GST_FLOW_OK);
|
|
|
|
if (rtp.buffer)
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
done:
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_vorbis_pay_handle_buffer (GstRTPBasePayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpVorbisPay *rtpvorbispay;
|
|
GstFlowReturn ret;
|
|
GstMapInfo map;
|
|
gsize size;
|
|
guint8 *data;
|
|
GstClockTime duration, timestamp;
|
|
guint8 VDT;
|
|
|
|
rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
data = map.data;
|
|
size = map.size;
|
|
duration = GST_BUFFER_DURATION (buffer);
|
|
timestamp = GST_BUFFER_PTS (buffer);
|
|
|
|
GST_LOG_OBJECT (rtpvorbispay, "size %" G_GSIZE_FORMAT
|
|
", duration %" GST_TIME_FORMAT, size, GST_TIME_ARGS (duration));
|
|
|
|
if (G_UNLIKELY (size < 1))
|
|
goto wrong_size;
|
|
|
|
/* find packet type */
|
|
if (data[0] & 1) {
|
|
/* header */
|
|
if (data[0] == 1) {
|
|
/* identification, we need to parse this in order to get the clock rate. */
|
|
if (G_UNLIKELY (!gst_rtp_vorbis_pay_parse_id (basepayload, data, size)))
|
|
goto parse_id_failed;
|
|
VDT = 1;
|
|
} else if (data[0] == 3) {
|
|
/* comment */
|
|
VDT = 2;
|
|
} else if (data[0] == 5) {
|
|
/* setup */
|
|
VDT = 1;
|
|
} else
|
|
goto unknown_header;
|
|
} else
|
|
/* data */
|
|
VDT = 0;
|
|
|
|
/* we need to collect the headers and construct a config string from them */
|
|
if (VDT != 0) {
|
|
GST_DEBUG_OBJECT (rtpvorbispay, "collecting header");
|
|
/* append header to the list of headers */
|
|
gst_buffer_unmap (buffer, &map);
|
|
rtpvorbispay->headers = g_list_append (rtpvorbispay->headers, buffer);
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
} else if (rtpvorbispay->headers && rtpvorbispay->need_headers) {
|
|
if (!gst_rtp_vorbis_pay_finish_headers (basepayload))
|
|
goto header_error;
|
|
}
|
|
|
|
/* there is a config request, see if we need to insert it */
|
|
if (rtpvorbispay->config_interval > 0 && rtpvorbispay->config_data) {
|
|
gboolean send_config = FALSE;
|
|
|
|
if (rtpvorbispay->last_config != -1) {
|
|
guint64 diff;
|
|
|
|
GST_LOG_OBJECT (rtpvorbispay,
|
|
"now %" GST_TIME_FORMAT ", last config %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (timestamp), GST_TIME_ARGS (rtpvorbispay->last_config));
|
|
|
|
/* calculate diff between last config in milliseconds */
|
|
if (timestamp > rtpvorbispay->last_config) {
|
|
diff = timestamp - rtpvorbispay->last_config;
|
|
} else {
|
|
diff = 0;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (rtpvorbispay,
|
|
"interval since last config %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
|
|
|
|
/* bigger than interval, queue config */
|
|
/* FIXME should convert timestamps to running time */
|
|
if (GST_TIME_AS_SECONDS (diff) >= rtpvorbispay->config_interval) {
|
|
GST_DEBUG_OBJECT (rtpvorbispay, "time to send config");
|
|
send_config = TRUE;
|
|
}
|
|
} else {
|
|
/* no known previous config time, send now */
|
|
GST_DEBUG_OBJECT (rtpvorbispay, "no previous config time, send now");
|
|
send_config = TRUE;
|
|
}
|
|
|
|
if (send_config) {
|
|
/* we need to send config now first */
|
|
/* different TDT type forces flush */
|
|
gst_rtp_vorbis_pay_payload_buffer (rtpvorbispay, 1,
|
|
NULL, rtpvorbispay->config_data, rtpvorbispay->config_size,
|
|
timestamp, GST_CLOCK_TIME_NONE, rtpvorbispay->config_extra_len);
|
|
|
|
if (timestamp != -1) {
|
|
rtpvorbispay->last_config = timestamp;
|
|
}
|
|
}
|
|
}
|
|
|
|
ret =
|
|
gst_rtp_vorbis_pay_payload_buffer (rtpvorbispay, VDT, buffer, data, size,
|
|
timestamp, duration, 0);
|
|
|
|
gst_buffer_unmap (buffer, &map);
|
|
gst_buffer_unref (buffer);
|
|
|
|
done:
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
wrong_size:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
|
|
("Invalid packet size (1 < %" G_GSIZE_FORMAT ")", size), (NULL));
|
|
gst_buffer_unmap (buffer, &map);
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
parse_id_failed:
|
|
{
|
|
gst_buffer_unmap (buffer, &map);
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
unknown_header:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
|
|
(NULL), ("Ignoring unknown header received"));
|
|
gst_buffer_unmap (buffer, &map);
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
header_error:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
|
|
(NULL), ("Error initializing header config"));
|
|
gst_buffer_unmap (buffer, &map);
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_vorbis_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
|
|
{
|
|
GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (payload);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_rtp_vorbis_pay_clear_packet (rtpvorbispay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
/* false to let parent handle event as well */
|
|
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_vorbis_pay_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstRtpVorbisPay *rtpvorbispay;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtpvorbispay = GST_RTP_VORBIS_PAY (element);
|
|
|
|
switch (transition) {
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_rtp_vorbis_pay_cleanup (rtpvorbispay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_vorbis_pay_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpVorbisPay *rtpvorbispay;
|
|
|
|
rtpvorbispay = GST_RTP_VORBIS_PAY (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_CONFIG_INTERVAL:
|
|
rtpvorbispay->config_interval = g_value_get_uint (value);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_vorbis_pay_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpVorbisPay *rtpvorbispay;
|
|
|
|
rtpvorbispay = GST_RTP_VORBIS_PAY (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_CONFIG_INTERVAL:
|
|
g_value_set_uint (value, rtpvorbispay->config_interval);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_vorbis_pay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpvorbispay",
|
|
GST_RANK_SECONDARY, GST_TYPE_RTP_VORBIS_PAY);
|
|
}
|