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Every g_quark_from_static_string() is a hash table lookup serialised on the global quark lock in GLib. Let's just look up the two quarks we need once and cache them locally for future use. While we're at it, add new utility functions for the two most commonly used tags (audio + video). Make first argument a gpointer so we don't have to cast and make the code ugly. These are used for logging purposes only anyway.
276 lines
8.1 KiB
C
276 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 "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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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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gboolean
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gst_rtp_celt_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpceltdepay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_DEPAY);
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}
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