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50f88db3ad
Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index): * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process): * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send): Fix some compiler warnings. Fixes #428182.
216 lines
6.3 KiB
C
216 lines
6.3 KiB
C
/* GStreamer
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* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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 "gstrtpspeexdepay.h"
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/* elementfactory information */
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static const GstElementDetails gst_rtp_speexdepay_details =
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GST_ELEMENT_DETAILS ("RTP packet depayloader",
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"Codec/Depayloader/Network",
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"Extracts Speex audio from RTP packets",
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"Edgard Lima <edgard.lima@indt.org.br>");
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/* RtpSPEEXDepay signals and args */
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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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ARG_0
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};
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static GstStaticPadTemplate gst_rtp_speex_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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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) [6000, 48000], "
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"encoding-name = (string) \"SPEEX\", "
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"encoding-params = (string) \"1\"")
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);
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static GstStaticPadTemplate gst_rtp_speex_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-speex")
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);
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static GstBuffer *gst_rtp_speex_depay_process (GstBaseRTPDepayload * depayload,
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GstBuffer * buf);
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static gboolean gst_rtp_speex_depay_setcaps (GstBaseRTPDepayload * depayload,
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GstCaps * caps);
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GST_BOILERPLATE (GstRtpSPEEXDepay, gst_rtp_speex_depay, GstBaseRTPDepayload,
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GST_TYPE_BASE_RTP_DEPAYLOAD);
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static void
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gst_rtp_speex_depay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_speex_depay_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_speex_depay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_speexdepay_details);
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}
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static void
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gst_rtp_speex_depay_class_init (GstRtpSPEEXDepayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
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gstbasertpdepayload_class->process = gst_rtp_speex_depay_process;
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gstbasertpdepayload_class->set_caps = gst_rtp_speex_depay_setcaps;
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}
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static void
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gst_rtp_speex_depay_init (GstRtpSPEEXDepay * rtpspeexdepay,
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GstRtpSPEEXDepayClass * klass)
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{
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GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay)->clock_rate = 8000;
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}
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static gint
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gst_rtp_speex_depay_get_mode (gint rate)
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{
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if (rate > 25000)
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return 2;
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else if (rate > 12500)
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return 1;
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else
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return 0;
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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_speex_comment[] =
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"\045\0\0\0Depayloaded with GStreamer speexdepay\0\0\0\0";
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static gboolean
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gst_rtp_speex_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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GstRtpSPEEXDepay *rtpspeexdepay;
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gint clock_rate, nb_channels;
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GstBuffer *buf;
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guint8 *data;
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const gchar *params;
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rtpspeexdepay = GST_RTP_SPEEX_DEPAY (depayload);
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "clock-rate", &clock_rate);
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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 = 1;
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else {
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nb_channels = atoi (params);
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}
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/* construct minimal header and comment packet for the decoder */
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buf = gst_buffer_new_and_alloc (80);
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data = GST_BUFFER_DATA (buf);
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memcpy (data, "Speex ", 8);
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data += 8;
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memcpy (data, "1.1.12", 7);
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data += 20;
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GST_WRITE_UINT32_LE (data, 1); /* version */
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data += 4;
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GST_WRITE_UINT32_LE (data, 80); /* header_size */
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data += 4;
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GST_WRITE_UINT32_LE (data, clock_rate); /* rate */
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data += 4;
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GST_WRITE_UINT32_LE (data, gst_rtp_speex_depay_get_mode (clock_rate)); /* mode */
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data += 4;
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GST_WRITE_UINT32_LE (data, 4); /* mode_bitstream_version */
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data += 4;
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GST_WRITE_UINT32_LE (data, nb_channels); /* nb_channels */
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data += 4;
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GST_WRITE_UINT32_LE (data, -1); /* bitrate */
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data += 4;
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GST_WRITE_UINT32_LE (data, 0xa0); /* frame_size */
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data += 4;
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GST_WRITE_UINT32_LE (data, 0); /* VBR */
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data += 4;
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GST_WRITE_UINT32_LE (data, 1); /* frames_per_packet */
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data += 4;
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GST_WRITE_UINT32_LE (data, 0); /* extra_headers */
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data += 4;
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GST_WRITE_UINT32_LE (data, 0); /* reserved1 */
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data += 4;
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GST_WRITE_UINT32_LE (data, 0); /* reserved2 */
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gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf);
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buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_speex_comment));
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memcpy (GST_BUFFER_DATA (buf), gst_rtp_speex_comment,
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sizeof (gst_rtp_speex_comment));
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gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpspeexdepay), buf);
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return TRUE;
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}
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static GstBuffer *
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gst_rtp_speex_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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{
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GstBuffer *outbuf = NULL;
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GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
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GST_BUFFER_SIZE (buf),
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gst_rtp_buffer_get_marker (buf),
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gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
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/* nothing special to be done */
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outbuf = gst_rtp_buffer_get_payload_buffer (buf);
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return outbuf;
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}
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gboolean
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gst_rtp_speex_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpspeexdepay",
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GST_RANK_MARGINAL, GST_TYPE_RTP_SPEEX_DEPAY);
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}
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