| /* GStreamer |
| * Copyright (C) <2007> Wim Taymans <wim.taymans@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/audio/audio.h> |
| |
| #include "gstrtpg722depay.h" |
| #include "gstrtpchannels.h" |
| #include "gstrtputils.h" |
| |
| GST_DEBUG_CATEGORY_STATIC (rtpg722depay_debug); |
| #define GST_CAT_DEFAULT (rtpg722depay_debug) |
| |
| static GstStaticPadTemplate gst_rtp_g722_depay_src_template = |
| GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/G722, " |
| "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") |
| ); |
| |
| static GstStaticPadTemplate gst_rtp_g722_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) 8000, " |
| /* "channels = (int) [1, MAX]" */ |
| /* "channel-order = (string) ANY" */ |
| "encoding-name = (string) \"G722\";" |
| "application/x-rtp, " |
| "media = (string) \"audio\", " |
| "payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", " |
| "clock-rate = (int) [ 1, MAX ]" |
| /* "channels = (int) [1, MAX]" */ |
| /* "emphasis = (string) ANY" */ |
| /* "channel-order = (string) ANY" */ |
| ) |
| ); |
| |
| #define gst_rtp_g722_depay_parent_class parent_class |
| G_DEFINE_TYPE (GstRtpG722Depay, gst_rtp_g722_depay, |
| GST_TYPE_RTP_BASE_DEPAYLOAD); |
| |
| static gboolean gst_rtp_g722_depay_setcaps (GstRTPBaseDepayload * depayload, |
| GstCaps * caps); |
| static GstBuffer *gst_rtp_g722_depay_process (GstRTPBaseDepayload * depayload, |
| GstRTPBuffer * rtp); |
| |
| static void |
| gst_rtp_g722_depay_class_init (GstRtpG722DepayClass * klass) |
| { |
| GstElementClass *gstelement_class; |
| GstRTPBaseDepayloadClass *gstrtpbasedepayload_class; |
| |
| GST_DEBUG_CATEGORY_INIT (rtpg722depay_debug, "rtpg722depay", 0, |
| "G722 RTP Depayloader"); |
| |
| gstelement_class = (GstElementClass *) klass; |
| gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass; |
| |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_g722_depay_src_template); |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_g722_depay_sink_template); |
| |
| gst_element_class_set_static_metadata (gstelement_class, |
| "RTP audio depayloader", "Codec/Depayloader/Network/RTP", |
| "Extracts G722 audio from RTP packets", |
| "Wim Taymans <wim.taymans@gmail.com>"); |
| |
| gstrtpbasedepayload_class->set_caps = gst_rtp_g722_depay_setcaps; |
| gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_g722_depay_process; |
| } |
| |
| static void |
| gst_rtp_g722_depay_init (GstRtpG722Depay * rtpg722depay) |
| { |
| } |
| |
| static gint |
| gst_rtp_g722_depay_parse_int (GstStructure * structure, const gchar * field, |
| gint def) |
| { |
| const gchar *str; |
| gint res; |
| |
| if ((str = gst_structure_get_string (structure, field))) |
| return atoi (str); |
| |
| if (gst_structure_get_int (structure, field, &res)) |
| return res; |
| |
| return def; |
| } |
| |
| static gboolean |
| gst_rtp_g722_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) |
| { |
| GstStructure *structure; |
| GstRtpG722Depay *rtpg722depay; |
| gint clock_rate, payload, samplerate; |
| gint channels; |
| GstCaps *srccaps; |
| gboolean res; |
| #if 0 |
| const gchar *channel_order; |
| const GstRTPChannelOrder *order; |
| #endif |
| |
| rtpg722depay = GST_RTP_G722_DEPAY (depayload); |
| |
| structure = gst_caps_get_structure (caps, 0); |
| |
| payload = 96; |
| gst_structure_get_int (structure, "payload", &payload); |
| switch (payload) { |
| case GST_RTP_PAYLOAD_G722: |
| channels = 1; |
| clock_rate = 8000; |
| samplerate = 16000; |
| break; |
| default: |
| /* no fixed mapping, we need clock-rate */ |
| channels = 0; |
| clock_rate = 0; |
| samplerate = 0; |
| break; |
| } |
| |
| /* caps can overwrite defaults */ |
| clock_rate = |
| gst_rtp_g722_depay_parse_int (structure, "clock-rate", clock_rate); |
| if (clock_rate == 0) |
| goto no_clockrate; |
| |
| if (clock_rate == 8000) |
| samplerate = 16000; |
| |
| if (samplerate == 0) |
| samplerate = clock_rate; |
| |
| channels = |
| gst_rtp_g722_depay_parse_int (structure, "encoding-params", channels); |
| if (channels == 0) { |
| channels = gst_rtp_g722_depay_parse_int (structure, "channels", channels); |
| if (channels == 0) { |
| /* channels defaults to 1 otherwise */ |
| channels = 1; |
| } |
| } |
| |
| depayload->clock_rate = clock_rate; |
| rtpg722depay->rate = samplerate; |
| rtpg722depay->channels = channels; |
| |
| srccaps = gst_caps_new_simple ("audio/G722", |
| "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL); |
| |
| /* FIXME: Do something with the channel order */ |
| #if 0 |
| /* add channel positions */ |
| channel_order = gst_structure_get_string (structure, "channel-order"); |
| |
| order = gst_rtp_channels_get_by_order (channels, channel_order); |
| if (order) { |
| gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), |
| order->pos); |
| } else { |
| GstAudioChannelPosition *pos; |
| |
| GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE, |
| (NULL), ("Unknown channel order '%s' for %d channels", |
| GST_STR_NULL (channel_order), channels)); |
| /* create default NONE layout */ |
| pos = gst_rtp_channels_create_default (channels); |
| gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos); |
| g_free (pos); |
| } |
| #endif |
| |
| res = gst_pad_set_caps (depayload->srcpad, srccaps); |
| gst_caps_unref (srccaps); |
| |
| return res; |
| |
| /* ERRORS */ |
| no_clockrate: |
| { |
| GST_ERROR_OBJECT (depayload, "no clock-rate specified"); |
| return FALSE; |
| } |
| } |
| |
| static GstBuffer * |
| gst_rtp_g722_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp) |
| { |
| GstRtpG722Depay *rtpg722depay; |
| GstBuffer *outbuf; |
| gint payload_len; |
| gboolean marker; |
| |
| rtpg722depay = GST_RTP_G722_DEPAY (depayload); |
| |
| payload_len = gst_rtp_buffer_get_payload_len (rtp); |
| |
| if (payload_len <= 0) |
| goto empty_packet; |
| |
| GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len); |
| |
| outbuf = gst_rtp_buffer_get_payload_buffer (rtp); |
| marker = gst_rtp_buffer_get_marker (rtp); |
| |
| if (marker && outbuf) { |
| /* mark talk spurt with RESYNC */ |
| GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC); |
| } |
| |
| if (outbuf) { |
| gst_rtp_drop_non_audio_meta (rtpg722depay, outbuf); |
| } |
| |
| return outbuf; |
| |
| /* ERRORS */ |
| empty_packet: |
| { |
| GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE, |
| ("Empty Payload."), (NULL)); |
| return NULL; |
| } |
| } |
| |
| gboolean |
| gst_rtp_g722_depay_plugin_init (GstPlugin * plugin) |
| { |
| return gst_element_register (plugin, "rtpg722depay", |
| GST_RANK_SECONDARY, GST_TYPE_RTP_G722_DEPAY); |
| } |