| /* GStreamer |
| * Copyright (C) <2005> Edgard Lima <edgard.lima@gmail.com> |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Library General Public |
| * License as published by the Free Software Foundation; either |
| * version 2 of the License, or (at your option) any later version. |
| * |
| * This library is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Library General Public License for more details. |
| * |
| * You should have received a copy of the GNU Library General Public |
| * License along with this library; if not, write to the |
| * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, USA. |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| # include "config.h" |
| #endif |
| |
| #include <string.h> |
| #include <stdlib.h> |
| #include <gst/rtp/gstrtpbuffer.h> |
| #include <gst/audio/audio.h> |
| |
| #include "gstrtpspeexdepay.h" |
| #include "gstrtputils.h" |
| |
| /* RtpSPEEXDepay signals and args */ |
| enum |
| { |
| /* FILL ME */ |
| LAST_SIGNAL |
| }; |
| |
| enum |
| { |
| PROP_0 |
| }; |
| |
| static GstStaticPadTemplate gst_rtp_speex_depay_sink_template = |
| GST_STATIC_PAD_TEMPLATE ("sink", |
| GST_PAD_SINK, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("application/x-rtp, " |
| "media = (string) \"audio\", " |
| "clock-rate = (int) [6000, 48000], " |
| "encoding-name = (string) \"SPEEX\"") |
| /* "encoding-params = (string) \"1\"" */ |
| ); |
| |
| static GstStaticPadTemplate gst_rtp_speex_depay_src_template = |
| GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-speex") |
| ); |
| |
| static GstBuffer *gst_rtp_speex_depay_process (GstRTPBaseDepayload * depayload, |
| GstRTPBuffer * rtp); |
| static gboolean gst_rtp_speex_depay_setcaps (GstRTPBaseDepayload * depayload, |
| GstCaps * caps); |
| |
| G_DEFINE_TYPE (GstRtpSPEEXDepay, gst_rtp_speex_depay, |
| GST_TYPE_RTP_BASE_DEPAYLOAD); |
| |
| static void |
| gst_rtp_speex_depay_class_init (GstRtpSPEEXDepayClass * klass) |
| { |
| GstElementClass *gstelement_class; |
| GstRTPBaseDepayloadClass *gstrtpbasedepayload_class; |
| |
| gstelement_class = (GstElementClass *) klass; |
| gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass; |
| |
| gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_speex_depay_process; |
| gstrtpbasedepayload_class->set_caps = gst_rtp_speex_depay_setcaps; |
| |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_speex_depay_src_template); |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_speex_depay_sink_template); |
| gst_element_class_set_static_metadata (gstelement_class, |
| "RTP Speex depayloader", "Codec/Depayloader/Network/RTP", |
| "Extracts Speex audio from RTP packets", |
| "Edgard Lima <edgard.lima@gmail.com>"); |
| } |
| |
| static void |
| gst_rtp_speex_depay_init (GstRtpSPEEXDepay * rtpspeexdepay) |
| { |
| } |
| |
| static gint |
| gst_rtp_speex_depay_get_mode (gint rate) |
| { |
| if (rate > 25000) |
| return 2; |
| else if (rate > 12500) |
| return 1; |
| else |
| return 0; |
| } |
| |
| /* len 4 bytes LE, |
| * vendor string (len bytes), |
| * user_len 4 (0) bytes LE |
| */ |
| static const gchar gst_rtp_speex_comment[] = |
| "\045\0\0\0Depayloaded with GStreamer speexdepay\0\0\0\0"; |
| |
| static gboolean |
| gst_rtp_speex_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) |
| { |
| GstStructure *structure; |
| GstRtpSPEEXDepay *rtpspeexdepay; |
| gint clock_rate, nb_channels; |
| GstBuffer *buf; |
| GstMapInfo map; |
| guint8 *data; |
| const gchar *params; |
| GstCaps *srccaps; |
| gboolean res; |
| |
| rtpspeexdepay = GST_RTP_SPEEX_DEPAY (depayload); |
| |
| structure = gst_caps_get_structure (caps, 0); |
| |
| if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) |
| goto no_clockrate; |
| depayload->clock_rate = clock_rate; |
| |
| if (!(params = gst_structure_get_string (structure, "encoding-params"))) |
| nb_channels = 1; |
| else { |
| nb_channels = atoi (params); |
| } |
| |
| /* construct minimal header and comment packet for the decoder */ |
| buf = gst_buffer_new_and_alloc (80); |
| gst_buffer_map (buf, &map, GST_MAP_WRITE); |
| data = map.data; |
| memcpy (data, "Speex ", 8); |
| data += 8; |
| memcpy (data, "1.1.12", 7); |
| data += 20; |
| GST_WRITE_UINT32_LE (data, 1); /* version */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, 80); /* header_size */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, clock_rate); /* rate */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, gst_rtp_speex_depay_get_mode (clock_rate)); /* mode */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, 4); /* mode_bitstream_version */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, nb_channels); /* nb_channels */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, -1); /* bitrate */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, 0xa0); /* frame_size */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, 0); /* VBR */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, 1); /* frames_per_packet */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, 0); /* extra_headers */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, 0); /* reserved1 */ |
| data += 4; |
| GST_WRITE_UINT32_LE (data, 0); /* reserved2 */ |
| gst_buffer_unmap (buf, &map); |
| |
| srccaps = gst_caps_new_empty_simple ("audio/x-speex"); |
| res = gst_pad_set_caps (depayload->srcpad, srccaps); |
| gst_caps_unref (srccaps); |
| |
| gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpspeexdepay), buf); |
| |
| buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_speex_comment)); |
| gst_buffer_fill (buf, 0, gst_rtp_speex_comment, |
| sizeof (gst_rtp_speex_comment)); |
| |
| gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpspeexdepay), buf); |
| |
| return res; |
| |
| /* ERRORS */ |
| no_clockrate: |
| { |
| GST_DEBUG_OBJECT (depayload, "no clock-rate specified"); |
| return FALSE; |
| } |
| } |
| |
| static GstBuffer * |
| gst_rtp_speex_depay_process (GstRTPBaseDepayload * depayload, |
| GstRTPBuffer * rtp) |
| { |
| GstBuffer *outbuf = NULL; |
| |
| GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d", |
| gst_buffer_get_size (rtp->buffer), |
| gst_rtp_buffer_get_marker (rtp), |
| gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp)); |
| |
| /* nothing special to be done */ |
| outbuf = gst_rtp_buffer_get_payload_buffer (rtp); |
| |
| if (outbuf) { |
| GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND; |
| gst_rtp_drop_non_audio_meta (depayload, outbuf); |
| } |
| |
| return outbuf; |
| } |
| |
| gboolean |
| gst_rtp_speex_depay_plugin_init (GstPlugin * plugin) |
| { |
| return gst_element_register (plugin, "rtpspeexdepay", |
| GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_DEPAY); |
| } |