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80f8780e92
Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
271 lines
8.1 KiB
C
271 lines
8.1 KiB
C
/* GStreamer
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* Copyright (C) <2009> 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 <stdlib.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/audio/audio.h>
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#include "gstrtpelements.h"
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#include "gstrtpceltdepay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpceltdepay_debug);
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#define GST_CAT_DEFAULT (rtpceltdepay_debug)
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/* RtpCELTDepay signals and args */
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#define DEFAULT_FRAMESIZE 480
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#define DEFAULT_CHANNELS 1
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#define DEFAULT_CLOCKRATE 32000
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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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 GstStaticPadTemplate gst_rtp_celt_depay_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 ("application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) [32000, 48000], "
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"encoding-name = (string) \"CELT\"")
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);
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static GstStaticPadTemplate gst_rtp_celt_depay_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 ("audio/x-celt")
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);
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static GstBuffer *gst_rtp_celt_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp);
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static gboolean gst_rtp_celt_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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#define gst_rtp_celt_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpCELTDepay, gst_rtp_celt_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpceltdepay, "rtpceltdepay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_DEPAY, rtp_element_init (plugin));
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static void
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gst_rtp_celt_depay_class_init (GstRtpCELTDepayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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GST_DEBUG_CATEGORY_INIT (rtpceltdepay_debug, "rtpceltdepay", 0,
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"CELT RTP Depayloader");
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_celt_depay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_celt_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP CELT depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts CELT audio from RTP packets",
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"Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_celt_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_celt_depay_setcaps;
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}
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static void
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gst_rtp_celt_depay_init (GstRtpCELTDepay * rtpceltdepay)
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{
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}
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/* len 4 bytes LE,
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* vendor string (len bytes),
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* user_len 4 (0) bytes LE
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*/
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static const gchar gst_rtp_celt_comment[] =
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"\045\0\0\0Depayloaded with GStreamer celtdepay\0\0\0\0";
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static gboolean
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gst_rtp_celt_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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GstRtpCELTDepay *rtpceltdepay;
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gint clock_rate, nb_channels = 0, frame_size = 0;
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GstBuffer *buf;
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GstMapInfo map;
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guint8 *ptr;
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const gchar *params;
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GstCaps *srccaps;
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gboolean res;
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rtpceltdepay = GST_RTP_CELT_DEPAY (depayload);
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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goto no_clockrate;
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depayload->clock_rate = clock_rate;
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if ((params = gst_structure_get_string (structure, "encoding-params")))
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nb_channels = atoi (params);
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if (!nb_channels)
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nb_channels = DEFAULT_CHANNELS;
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if ((params = gst_structure_get_string (structure, "frame-size")))
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frame_size = atoi (params);
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if (!frame_size)
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frame_size = DEFAULT_FRAMESIZE;
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rtpceltdepay->frame_size = frame_size;
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GST_DEBUG_OBJECT (depayload, "clock-rate=%d channels=%d frame-size=%d",
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clock_rate, nb_channels, frame_size);
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/* construct minimal header and comment packet for the decoder */
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buf = gst_buffer_new_and_alloc (60);
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gst_buffer_map (buf, &map, GST_MAP_WRITE);
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ptr = map.data;
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memcpy (ptr, "CELT ", 8);
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ptr += 8;
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memcpy (ptr, "1.1.12", 7);
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ptr += 20;
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GST_WRITE_UINT32_LE (ptr, 0x80000006); /* version */
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ptr += 4;
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GST_WRITE_UINT32_LE (ptr, 56); /* header_size */
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ptr += 4;
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GST_WRITE_UINT32_LE (ptr, clock_rate); /* rate */
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ptr += 4;
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GST_WRITE_UINT32_LE (ptr, nb_channels); /* channels */
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ptr += 4;
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GST_WRITE_UINT32_LE (ptr, frame_size); /* frame-size */
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ptr += 4;
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GST_WRITE_UINT32_LE (ptr, -1); /* overlap */
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ptr += 4;
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GST_WRITE_UINT32_LE (ptr, -1); /* bytes_per_packet */
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ptr += 4;
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GST_WRITE_UINT32_LE (ptr, 0); /* extra headers */
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gst_buffer_unmap (buf, &map);
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srccaps = gst_caps_new_empty_simple ("audio/x-celt");
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res = gst_pad_set_caps (depayload->srcpad, srccaps);
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gst_caps_unref (srccaps);
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gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpceltdepay), buf);
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buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_celt_comment));
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gst_buffer_fill (buf, 0, gst_rtp_celt_comment, sizeof (gst_rtp_celt_comment));
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gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpceltdepay), buf);
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return res;
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/* ERRORS */
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no_clockrate:
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{
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GST_ERROR_OBJECT (depayload, "no clock-rate specified");
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return FALSE;
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}
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}
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static GstBuffer *
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gst_rtp_celt_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
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GstBuffer *outbuf = NULL;
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guint8 *payload;
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guint offset, pos, payload_len, total_size, size;
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guint8 s;
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gint clock_rate = 0, frame_size = 0;
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GstClockTime framesize_ns = 0, timestamp;
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guint n = 0;
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GstRtpCELTDepay *rtpceltdepay;
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rtpceltdepay = GST_RTP_CELT_DEPAY (depayload);
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clock_rate = depayload->clock_rate;
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frame_size = rtpceltdepay->frame_size;
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framesize_ns = gst_util_uint64_scale_int (frame_size, GST_SECOND, clock_rate);
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timestamp = GST_BUFFER_PTS (rtp->buffer);
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GST_LOG_OBJECT (depayload,
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"got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
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gst_buffer_get_size (rtp->buffer), gst_rtp_buffer_get_marker (rtp),
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gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
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GST_LOG_OBJECT (depayload, "got clock-rate=%d, frame_size=%d, "
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"_ns=%" GST_TIME_FORMAT ", timestamp=%" GST_TIME_FORMAT, clock_rate,
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frame_size, GST_TIME_ARGS (framesize_ns), GST_TIME_ARGS (timestamp));
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payload = gst_rtp_buffer_get_payload (rtp);
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payload_len = gst_rtp_buffer_get_payload_len (rtp);
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/* first count how many bytes are consumed by the size headers and make offset
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* point to the first data byte */
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total_size = 0;
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offset = 0;
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while (total_size < payload_len) {
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do {
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s = payload[offset++];
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total_size += s + 1;
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} while (s == 0xff);
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}
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/* offset is now pointing to the payload */
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total_size = 0;
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pos = 0;
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while (total_size < payload_len) {
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n++;
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size = 0;
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do {
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s = payload[pos++];
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size += s;
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total_size += s + 1;
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} while (s == 0xff);
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outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, offset, size);
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offset += size;
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if (frame_size != -1 && clock_rate != -1) {
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GST_BUFFER_PTS (outbuf) = timestamp + framesize_ns * n;
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GST_BUFFER_DURATION (outbuf) = framesize_ns;
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}
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GST_LOG_OBJECT (depayload, "push timestamp=%"
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GST_TIME_FORMAT ", duration=%" GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
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gst_rtp_drop_non_audio_meta (depayload, outbuf);
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gst_rtp_base_depayload_push (depayload, outbuf);
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}
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return NULL;
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}
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