From 40cbc01b87285a52f333beba60074075db17eaa5 Mon Sep 17 00:00:00 2001 From: Anders Broman Date: Tue, 27 Jun 2006 21:44:07 +0000 Subject: [PATCH] From Alejandro Vaquero: The "listen_rtp" plugin that allows to listen audio RTP conversations. The plugin is integrated to the "Voip Calls" feature. There is a new "listen" button in the "Voip Calls" that once the calls are selected and the "listen" is clicked, a new window will open. In this window you can change the simulated jitter buffer to be used for decoding the RTP packets. In this first implementation, only a static jitter buffer can be simulated. Then pressing "decode" will decode all the RTP and generate the graphical view of the audio channels. From there you can select up to two channels at the same time (to be played in the left and right channels) and then play, pause, stop,etc.....ok, the attached screenshot it is self explanatory (hopefully). The only codecs available from now are G711u and G711a law. The PortAudio (www.portaudio.com) library is used to play audio. This is an open source cross-platform Audio library. I have tested this on Windows XP and in a linux FC4. NOTE: only the plugin soure files checked in at this time. svn path=/trunk/; revision=18588 --- plugins/listen_rtp/AUTHORS | 5 + plugins/listen_rtp/Makefile.am | 54 + plugins/listen_rtp/Makefile.nmake | 53 + plugins/listen_rtp/NEWS | 1 + plugins/listen_rtp/codecs/G711a/G711adecode.c | 18 + plugins/listen_rtp/codecs/G711a/G711adecode.h | 3 + plugins/listen_rtp/codecs/G711a/G711atable.h | 36 + plugins/listen_rtp/codecs/G711u/G711udecode.c | 18 + plugins/listen_rtp/codecs/G711u/G711udecode.h | 3 + plugins/listen_rtp/codecs/G711u/G711utable.h | 36 + plugins/listen_rtp/listen_rtp_plugin.c | 49 + plugins/listen_rtp/main_listen_rtp.c | 1857 +++++++++++++++++ plugins/listen_rtp/main_listen_rtp.h | 29 + 13 files changed, 2162 insertions(+) create mode 100644 plugins/listen_rtp/AUTHORS create mode 100644 plugins/listen_rtp/Makefile.am create mode 100644 plugins/listen_rtp/Makefile.nmake create mode 100644 plugins/listen_rtp/NEWS create mode 100644 plugins/listen_rtp/codecs/G711a/G711adecode.c create mode 100644 plugins/listen_rtp/codecs/G711a/G711adecode.h create mode 100644 plugins/listen_rtp/codecs/G711a/G711atable.h create mode 100644 plugins/listen_rtp/codecs/G711u/G711udecode.c create mode 100644 plugins/listen_rtp/codecs/G711u/G711udecode.h create mode 100644 plugins/listen_rtp/codecs/G711u/G711utable.h create mode 100644 plugins/listen_rtp/listen_rtp_plugin.c create mode 100644 plugins/listen_rtp/main_listen_rtp.c create mode 100644 plugins/listen_rtp/main_listen_rtp.h diff --git a/plugins/listen_rtp/AUTHORS b/plugins/listen_rtp/AUTHORS new file mode 100644 index 0000000000..ecabd2824f --- /dev/null +++ b/plugins/listen_rtp/AUTHORS @@ -0,0 +1,5 @@ +$Id$ + +Author: +Alejandro Vaquero + diff --git a/plugins/listen_rtp/Makefile.am b/plugins/listen_rtp/Makefile.am new file mode 100644 index 0000000000..a568eaad6c --- /dev/null +++ b/plugins/listen_rtp/Makefile.am @@ -0,0 +1,54 @@ +# Makefile.am +# Automake file for listen_rtp Ethereal plugin +# +# $Id$ +# +# Ethereal - Network traffic analyzer +# By Gerald Combs +# Copyright 1998 Gerald Combs +# +# This program is free software; you can redistribute it and/or +# modify it under the terms of the GNU General Public License +# as published by the Free Software Foundation; either version 2 +# of the License, or (at your option) any later version. +# +# This program 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 General Public License for more details. +# +# You should have received a copy of the GNU General Public License +# along with this program; if not, write to the Free Software +# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. +# + +INCLUDES = -I$(top_srcdir) @PORTAUDIO_INCLUDES@ + +plugindir = @plugindir@ + +plugin_LTLIBRARIES = listen_rtp.la + +listen_rtp_la_SOURCES = \ + listen_rtp_plugin.c \ + main_listen_rtp.c \ + main_listen_rtp.h \ + codecs/G711a/G711adecode.c codecs/G711a/G711adecode.h \ + codecs/G711u/G711udecode.c codecs/G711u/G711udecode.h + +listen_rtp_la_LDFLAGS = -module -avoid-version +listen_rtp_la_LIBADD = @PLUGIN_LIBS@ @PORTAUDIO_LIBS@ + +# Libs must be cleared, or else libtool won't create a shared module. +# If your module needs to be linked against any particular libraries, +# add them here. +LIBS = + +CLEANFILES = \ + listen_rtp \ + *~ + +MAINTAINERCLEANFILES = \ + Makefile.in + +EXTRA_DIST = \ + Makefile.nmake diff --git a/plugins/listen_rtp/Makefile.nmake b/plugins/listen_rtp/Makefile.nmake new file mode 100644 index 0000000000..68debf962e --- /dev/null +++ b/plugins/listen_rtp/Makefile.nmake @@ -0,0 +1,53 @@ +# +# $Id$ +# + +include ..\..\config.nmake +include + +PA_CFLAGS=/I$(PORTAUDIO_DIR)\pa_common +WIN_LIBS=winmm.lib + +############### no need to modify below this line ######### + +CFLAGS=/DHAVE_CONFIG_H /I../.. /I../../wiretap $(GTK2_CFLAGS) $(GLIB_CFLAGS) \ + /I$(PCAP_DIR)\include -D_U_="" $(LOCAL_CFLAGS) $(PA_CFLAGS) + +PACFLAGS = -c /Zi /W1 /Od $(PA_CFLAGS) + +.c.obj:: + $(CC) $(CFLAGS) -Fdlisten_rtp.pdb -c $< + +LDFLAGS = /NOLOGO /INCREMENTAL:no /MACHINE:I386 $(LOCAL_LDFLAGS) + +!IFDEF ENABLE_LIBWIRESHARK +LINK_PLUGIN_WITH=..\..\epan\libwireshark.lib +CFLAGS=/DHAVE_WIN32_LIBWIRESHARK_LIB /D_NEED_VAR_IMPORT_ $(CFLAGS) + +OBJECTS=listen_rtp_plugin.obj main_listen_rtp.obj pa_lib.obj pa_win_wmme.obj \ + G711udecode.obj G711adecode.obj + +listen_rtp.dll listen_rtp.exp listen_rtp.lib : $(OBJECTS) $(LINK_PLUGIN_WITH) + link -dll /out:listen_rtp.dll $(LDFLAGS) $(OBJECTS) $(LINK_PLUGIN_WITH) \ + $(GLIB_LIBS) $(GTK2_LIBS) $(WIN_LIBS) + +!ENDIF + +pa_lib.obj: $(PORTAUDIO_DIR)\pa_common\pa_lib.c + $(CC) $(PACFLAGS) $(PORTAUDIO_DIR)\pa_common\pa_lib.c -o $@ + +pa_win_wmme.obj: $(PORTAUDIO_DIR)\pa_win_wmme\pa_win_wmme.c + $(CC) $(PACFLAGS) $(PORTAUDIO_DIR)\pa_win_wmme\pa_win_wmme.c -o $@ + +G711adecode.obj: codecs\G711a\G711adecode.c codecs\G711a\G711adecode.h codecs\G711a\G711atable.h + $(CC) $(CFLAGS) -c /Zi /W1 /Od codecs\G711a\G711adecode.c -o $@ + +G711udecode.obj: codecs\G711u\G711udecode.c codecs\G711u\G711udecode.h codecs\G711u\G711utable.h + $(CC) $(CFLAGS) -c /Zi /W1 /Od codecs\G711u\G711udecode.c -o $@ + +clean: + rm -f $(OBJECTS) listen_rtp.dll listen_rtp.exp listen_rtp.lib *.pdb + +distclean: clean + +maintainer-clean: distclean diff --git a/plugins/listen_rtp/NEWS b/plugins/listen_rtp/NEWS new file mode 100644 index 0000000000..055c8729cd --- /dev/null +++ b/plugins/listen_rtp/NEWS @@ -0,0 +1 @@ +$Id$ diff --git a/plugins/listen_rtp/codecs/G711a/G711adecode.c b/plugins/listen_rtp/codecs/G711a/G711adecode.c new file mode 100644 index 0000000000..644c24d9a9 --- /dev/null +++ b/plugins/listen_rtp/codecs/G711a/G711adecode.c @@ -0,0 +1,18 @@ +#include +#include "G711adecode.h" +#include "G711atable.h" + +int +decodeG711a(void *input, int inputSizeBytes, void *output, int *outputSizeBytes) +{ + guint8 *dataIn = (guint8 *)input; + gint16 *dataOut = (gint16 *)output; + int i; + + for (i=0; i +#include "G711udecode.h" +#include "G711utable.h" + +int +decodeG711u(void *input, int inputSizeBytes, void *output, int *outputSizeBytes) +{ + guint8 *dataIn = (guint8 *)input; + gint16 *dataOut = (gint16 *)output; + int i; + + for (i=0; i +* Copyright 1998 Gerald Combs +* +* This program is free software; you can redistribute it and/or +* modify it under the terms of the GNU General Public License +* as published by the Free Software Foundation; either version 2 +* of the License, or (at your option) any later version. +* +* This program 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 General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. +*/ + +#ifndef ENABLE_STATIC +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include + +#include + +#include "main_listen_rtp.h" + +G_MODULE_EXPORT const gchar version[] = "0.1"; + +G_MODULE_EXPORT void plugin_register(void) +{ + proto_register_listen_rtp(); +} + +G_MODULE_EXPORT void plugin_reg_handoff(void) +{ + proto_reg_handoff_listen_rtp(); +} +#endif + diff --git a/plugins/listen_rtp/main_listen_rtp.c b/plugins/listen_rtp/main_listen_rtp.c new file mode 100644 index 0000000000..a659687f95 --- /dev/null +++ b/plugins/listen_rtp/main_listen_rtp.c @@ -0,0 +1,1857 @@ +/* main_listen_rtp.c +* Listen RTP +* +* (c) 2006, Alejandro Vaquero +* +* Here is a summary on how this listen RTP works: +* - The VoipCalls will call add_rtp_packet() every time there is an RTP packet +* - add_rtp_packet() will add the RTP packet in a RTP stream struct, and create the RTP stream if it is the +* first RTP in the stream. +* - Each new RTP stream will be added to a list of RTP stream, called rtp_streams_list +* - When the user clicks "Listen" in the VoipCall dialoge, listen_rtp_init() is called. +* - listen_rtp_init() create the main dialog, and it calls: +* + mark_rtp_stream_to_play() to mark the RTP streams that needs to be displayed. These are the RTP stream +* that match the selected calls in the VoipCall dlg. +* + decode_rtp_stream() this will decode the RTP packets in each RTP stream, and will also create +* the RTP channles. An RTP channel is a group of RTP stream that have in common the source and destination +* IP and UPD ports. The RTP channels is what the user will listen in one of the two Audio channles. +* The RTP channels are stored in the hash table rtp_channels_hash +* + add_channel_to_window() will create and add the Audio graphic representation in the main window +* - When the user click the check box to listen one of the Audio channels, the structure rtp_channels is filled +* to play one or two RTP channels (a max of two channels can be listened at a given moment) +* +* +* +* $Id$ +* +* Wireshark - Network traffic analyzer +* By Gerald Combs +* Copyright 1999 Gerald Combs +* +* This program is free software; you can redistribute it and/or +* modify it under the terms of the GNU General Public License +* as published by the Free Software Foundation; either version 2 +* of the License, or (at your option) any later version. +* +* This program 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 General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include +#include "globals.h" +#include "portaudio.h" +#include "simple_dialog.h" +#include "../../gtk/gui_utils.h" +#include "../../gtk/dlg_utils.h" +#include "../../gtk/compat_macros.h" + +#include "../../gtk/graph_analysis.h" +#include "../../gtk/voip_calls_dlg.h" +#include "../../gtk/voip_calls.h" +#include "../../gtk/gtkglobals.h" + + +#include + +#include "main_listen_rtp.h" +#include "codecs/G711a/G711adecode.h" +#include "codecs/G711u/G711udecode.h" +#include "main_listen_rtp.h" +#include + +/*define this symbol to compile with G729 and G723 codecs*/ +/*#define HAVE_G729_G723 1*/ + +#ifdef HAVE_G729_G723 +#include "codecs/G729/G729decode.h" +#include "codecs/G723/G723decode.h" +#endif + +static gboolean initialized = FALSE; + +static int proto_listen_rtp = -1; +voip_calls_tapinfo_t *voip_calls = NULL; + +/* Hash table with all the RTP streams */ +static GHashTable* rtp_streams_hash = NULL; + +/* List with all the RTP streams (this is used to decode them as it is sorted)*/ +static GList* rtp_streams_list = NULL; + +/* the window */ +static GtkWidget *listen_rtp_dlg_w; +static GtkWidget *channels_vb; +static GtkWidget *main_scrolled_window = NULL; +static GtkWidget *jitter_spinner; +static GtkWidget *bt_decode; +static GtkWidget *bt_play; +static GtkWidget *bt_pause; +static GtkWidget *bt_stop; +static GtkWidget *progress_bar; +static GtkWidget *info_bar; +static GtkWidget *stat_hbox; + +static guint32 total_packets; +static guint32 total_frames; +static guint32 progbar_count; + +static int new_jitter_buff; + +/* a hash table with the RTP streams to play per audio channel */ +static GHashTable *rtp_channels_hash = NULL; + +/* Port Audio staff */ +#define SAMPLE_RATE (8000) +#define NUM_CHANNELS (2) + +#define PA_SAMPLE_TYPE paInt16 +typedef gint16 SAMPLE; +#define SAMPLE_SILENCE (0) +#define FRAMES_PER_BUFFER (512) + +typedef struct _sample_t { + SAMPLE val; + guint8 status; +} sample_t; + +#define S_NORMAL 0 +#define S_DROP_BY_JITT 1 +#define S_WRONG_SEQ 2 + +/* Display channels constants */ +#define MULT 80 +#define CHANNEL_WIDTH 500 +#define CHANNEL_HEIGHT 100 +#define MAX_TIME_LABEL 10 +#define HEIGHT_TIME_LABEL 18 +#define MAX_NUM_COL_CONV 10 + +PortAudioStream *pa_stream; + +/* defines a RTP stream */ +typedef struct _rtp_stream_info { + address src_addr; + guint16 src_port; + address dest_addr; + guint16 dest_port; + guint32 ssrc; + guint32 first_frame_number; /* first RTP frame for the stream */ + double start_time; /* RTP stream start time in ms */ + gboolean play; + guint16 call_num; + GList* rtp_packets_list; /* List of RTP packets in the stream */ + guint32 num_packets; +} rtp_stream_info_t; + + +/* defines the RTP streams to be played in an audio channel */ +typedef struct _rtp_channel_info { + double start_time; /* RTP stream start time in ms */ + double end_time; /* RTP stream end time in ms */ + GArray *samples; /* the array with decoded audio */ + guint16 call_num; + gboolean selected; + guint32 frame_index; + guint32 drop_by_jitter_buff; + guint32 out_of_seq; + guint32 max_frame_index; + GtkWidget *check_bt; + GtkWidget *separator; + GtkWidget *scroll_window; + GtkWidget *draw_area; + GdkPixmap *pixmap; + GtkAdjustment *h_scrollbar_adjustment; + GdkPixbuf* cursor_pixbuf; + PaTimestamp cursor_prev; + GdkGC *bg_gc[MAX_NUM_COL_CONV+1]; + gboolean cursor_catch; + rtp_stream_info_t *first_stream; /* This is the first RTP stream in the channel */ + guint32 num_packets; +} rtp_channel_info_t; + +/* defines a RTP packet */ +typedef struct _rtp_packet { + struct _rtp_info *info; /* the RTP dissected info */ + double arrive_offset; /* arrive offset time since the begining of the stream in ms */ +/* const guint8* payload_data; */ + guint8* payload_data; +} rtp_packet_t; + +/* defines the two RTP channels to be played */ +typedef struct _rtp_play_channles { + rtp_channel_info_t* rci[2]; /* Channels to be played */ + guint32 start_index[2]; + guint32 end_index[2]; + int channel; + guint32 max_frame_index; + guint32 frame_index; + gboolean pause; + gboolean stop; + gint32 pause_duration; + PaTimestamp out_diff_time; +} rtp_play_channles_t; + +/* The two RTP channles to play */ +static rtp_play_channles_t *rtp_channels = NULL; + + +/****************************************************************************/ +static void +rtp_key_destroy(gchar *key) +{ + g_free(key); + key = NULL; +} + +/****************************************************************************/ +static void +rtp_channel_value_destroy(rtp_channel_info_t *rci) +{ + g_array_free(rci->samples, TRUE); + g_free(rci); + rci = NULL; +} + +/****************************************************************************/ +static void +rtp_stream_value_destroy(rtp_stream_info_t *rsi) +{ + GList* rtp_packets_list; + rtp_packet_t *rp; + + rtp_packets_list = g_list_first(rsi->rtp_packets_list); + while (rtp_packets_list) + { + rp = rtp_packets_list->data; + + g_free(rp->info); + g_free(rp->payload_data); + g_free(rp); + rp = NULL; + + rtp_packets_list = g_list_next(rtp_packets_list); + } + g_free(rsi); + rsi = NULL; +} + +/****************************************************************************/ +static void +set_sensitive_check_bt(gchar *key _U_ , rtp_channel_info_t *rci, guint *stop _U_ ) +{ + gtk_widget_set_sensitive(rci->check_bt, !(*stop)); +} + +/****************************************************************************/ +static void +bt_state(gboolean decode, gboolean play, gboolean pause, gboolean stop) +{ + gboolean new_jitter_value = FALSE; + gboolean false_val = FALSE; + + gtk_widget_set_sensitive(bt_decode, decode); + gtk_widget_set_sensitive(jitter_spinner, decode); + + if (new_jitter_buff != (int) gtk_spin_button_get_value((GtkSpinButton * )jitter_spinner)) { + new_jitter_value = TRUE; + } + + /* set the sensitive state of play only if there is a channel selected */ + if ( play && (rtp_channels->rci[0] || rtp_channels->rci[1]) && !new_jitter_value) { + gtk_widget_set_sensitive(bt_play, TRUE); + } else { + gtk_widget_set_sensitive(bt_play, FALSE); + } + + if (!new_jitter_value) { + gtk_widget_set_sensitive(bt_pause, pause); + gtk_widget_set_sensitive(bt_stop, stop); + + /* Set sensitive to the check buttons based on the STOP state */ + g_hash_table_foreach( rtp_channels_hash, (GHFunc)set_sensitive_check_bt, &stop); + } else { + gtk_widget_set_sensitive(bt_pause, FALSE); + gtk_widget_set_sensitive(bt_stop, FALSE); + + g_hash_table_foreach( rtp_channels_hash, (GHFunc)set_sensitive_check_bt, &false_val); + } +} + +/****************************************************************************/ +static void +add_rtp_packet(struct _rtp_info *rtp_info, packet_info *pinfo) +{ + rtp_stream_info_t *stream_info = NULL; + rtp_packet_t *new_rtp_packet; + GString *key_str = NULL; + + /* create the the streams hash if it doen't exist */ + if (!rtp_streams_hash) + rtp_streams_hash = g_hash_table_new_full( g_str_hash, g_str_equal, rtp_key_destroy, rtp_stream_value_destroy); + + /* Create a hash key to lookup in the RTP streams hash table + * uses: src_ip:src_port dst_ip:dst_port ssrc + */ + key_str = g_string_new(""); + g_string_printf(key_str, "%s:%d %s:%d %d", get_addr_name(&(pinfo->src)), + pinfo->srcport, get_addr_name(&(pinfo->dst)), + pinfo->destport, rtp_info->info_sync_src ); + + /* lookup for this rtp packet in the stream hash table*/ + stream_info = g_hash_table_lookup( rtp_streams_hash, key_str->str); + + /* if it is not in the hash table, create a new stream */ + if (stream_info==NULL) { + stream_info = g_malloc(sizeof(rtp_stream_info_t)); + COPY_ADDRESS(&(stream_info->src_addr), &(pinfo->src)); + stream_info->src_port = pinfo->srcport; + COPY_ADDRESS(&(stream_info->dest_addr), &(pinfo->dst)); + stream_info->dest_port = pinfo->destport; + stream_info->ssrc = rtp_info->info_sync_src; + stream_info->rtp_packets_list = NULL; + stream_info->first_frame_number = pinfo->fd->num; + stream_info->start_time = nstime_to_msec(&pinfo->fd->rel_ts); + stream_info->call_num = 0; + stream_info->play = FALSE; + stream_info->num_packets = 0; + + g_hash_table_insert(rtp_streams_hash, g_strdup(key_str->str), stream_info); + + /* Add the element to the List too. The List is used to decode the packets because it is sordted */ + rtp_streams_list = g_list_append(rtp_streams_list, stream_info); + } + + /* increment the number of packets in this stream, this is used for the progress bar and statistics*/ + stream_info->num_packets++; + + /* Add the RTP packet to the list */ + new_rtp_packet = g_malloc(sizeof(rtp_packet_t)); + new_rtp_packet->info = g_malloc(sizeof(struct _rtp_info)); + + memcpy(new_rtp_packet->info, rtp_info, sizeof(struct _rtp_info)); + new_rtp_packet->arrive_offset = nstime_to_msec(&pinfo->fd->rel_ts) - stream_info->start_time; + /* copy the RTP payload to the rtp_packet to be decoded later */ + if (rtp_info->info_payload_len) { + new_rtp_packet->payload_data = g_malloc(rtp_info->info_payload_len); + memcpy(new_rtp_packet->payload_data, &(rtp_info->info_data[rtp_info->info_payload_offset]), rtp_info->info_payload_len); + } else { + new_rtp_packet->payload_data = NULL; + } + + stream_info->rtp_packets_list = g_list_append(stream_info->rtp_packets_list, new_rtp_packet); + + g_string_free(key_str, TRUE); +} + +/****************************************************************************/ +/* Mark the RTP stream to be played. Use the voip_calls graph to see if the + * setup_frame is there and then if the associated voip_call is selected. + */ +static void +mark_rtp_stream_to_play(gchar *key _U_ , rtp_stream_info_t *rsi, gpointer ptr _U_) +{ + GList* graph_list; + graph_analysis_item_t *graph_item; + GList* voip_calls_list; + voip_calls_info_t *tmp_voip_call; + + /* Reset the "to be play" value because the user can close and reopen the Listen RTP window + * and the streams are nor reset in that case + */ + rsi->play = FALSE; + + /* and associate the RTP stream with a call using the first RTP in the stream*/ + graph_list = g_list_first(voip_calls->graph_analysis->list); + while (graph_list) + { + graph_item = graph_list->data; + if (rsi->first_frame_number == graph_item->frame_num) { + rsi->call_num = graph_item->conv_num; + /* if it is in the graph list, then check if the voip_call is selected */ + voip_calls_list = g_list_first(voip_calls->strinfo_list); + while (voip_calls_list) + { + tmp_voip_call = voip_calls_list->data; + if ( (tmp_voip_call->call_num == rsi->call_num) && (tmp_voip_call->selected == TRUE) ) { + rsi->play = TRUE; + total_packets += rsi->num_packets; + break; + } + voip_calls_list = g_list_next(voip_calls_list); + } + break; + } + graph_list = g_list_next(graph_list); + } +} + + +/****************************************************************************/ +/* Decode a RTP packet + * Return the number of decoded bytes + */ +static int +decode_rtp_packet(rtp_packet_t *rp, rtp_channel_info_t *rci, SAMPLE **out_buff) +{ + unsigned int payload_type; + SAMPLE *tmp_buff = NULL; + int decoded_bytes = 0; + + if ((rp->payload_data == NULL) || (rp->info->info_payload_len == 0) ) { + return 0; + } + + payload_type = rp->info->info_payload_type; + switch (payload_type) { + case 0: /* G711 Ulaw */ + tmp_buff = malloc(sizeof(SAMPLE) * rp->info->info_payload_len * 1); + decodeG711u(rp->payload_data, rp->info->info_payload_len, + tmp_buff, &decoded_bytes); + break; + case 8: /* G711 Alaw */ + tmp_buff = malloc(sizeof(SAMPLE) * rp->info->info_payload_len * 1); + decodeG711a(rp->payload_data, rp->info->info_payload_len, + tmp_buff, &decoded_bytes); + break; +#ifdef HAVE_G729_G723 + case 18: /* G729 */ + tmp_buff = malloc(sizeof(SAMPLE) * rp->info->info_payload_len * 8); /* G729 8kbps => 64kbps/8kbps = 8 */ + decodeG729(rp->payload_data, rp->info->info_payload_len, + tmp_buff, &decoded_bytes); + break; + case 4: /* G723 */ + + if (rp->info->info_payload_len%24 == 0) /* G723 High 6.4kbps */ + tmp_buff = malloc(sizeof(SAMPLE) * rp->info->info_payload_len * 10); /* G723 High 64kbps/6.4kbps = 10 */ + else if (rp->info->info_payload_len%20 == 0) /* G723 Low 5.3kbps */ + tmp_buff = malloc(sizeof(SAMPLE) * rp->info->info_payload_len * 13); /* G723 High 64kbps/5.3kbps = 13 */ + else { + return 0; + } + decodeG723(rp->payload_data, rp->info->info_payload_len, + tmp_buff, &decoded_bytes); + break; +#endif + } + + *out_buff = tmp_buff; + return decoded_bytes; +} + +/****************************************************************************/ +void +update_progress_bar(gfloat percentage) +{ + + gtk_progress_bar_update(GTK_PROGRESS_BAR(progress_bar), percentage); + + /* Force gtk to redraw the window before starting decoding the packet */ + while (gtk_events_pending()) + gtk_main_iteration(); +} + +/****************************************************************************/ +/* Decode the RTP streams and add them to the RTP channels struct + */ +static void +decode_rtp_stream(rtp_stream_info_t *rsi, gpointer ptr _U_) +{ + GString *key_str = NULL; + rtp_channel_info_t *rci; + gboolean first = TRUE; + GList* rtp_packets_list; + rtp_packet_t *rp; + + int i; + double rtp_time; + double rtp_time_prev; + double arrive_time; + double arrive_time_prev; + double start_time; + double start_rtp_time; + double diff; + double pack_period; + double total_time; + double total_time_prev; + gint32 silence_frames; + int seq; + double delay; + double prev_diff; + double mean_delay; + double variation; + int decoded_bytes; + int decoded_bytes_prev; + int jitter_buff; + SAMPLE *out_buff = NULL; + sample_t silence; + sample_t sample; + guint8 status; + guint32 start_timestamp; + + guint32 progbar_nextstep; + int progbar_quantum; + gfloat progbar_val; + + silence.val = 0; + silence.status = S_NORMAL; + + /* skip it if we are not going to play it */ + if (rsi->play == FALSE) { + return; + } + + /* get the static jitter buffer from the spinner gui */ + jitter_buff = (int) gtk_spin_button_get_value((GtkSpinButton * )jitter_spinner); + + /* Create a hash key to lookup in the RTP channels hash + * uses: src_ip:src_port dst_ip:dst_port call_num + */ + key_str = g_string_new(""); + g_string_printf(key_str, "%s:%d %s:%d %d", get_addr_name(&(rsi->src_addr)), + rsi->src_port, get_addr_name(&(rsi->dest_addr)), + rsi->dest_port, rsi->call_num ); + + /* create the rtp_channels_hash table if it doesn't exist */ + if (!rtp_channels_hash) { + rtp_channels_hash = g_hash_table_new_full( g_str_hash, g_str_equal, rtp_key_destroy, rtp_channel_value_destroy); + } + + /* lookup for this stream in the channel hash table */ + rci = g_hash_table_lookup( rtp_channels_hash, key_str->str); + + /* ..if it is not in the hash, create an entry */ + if (rci == NULL) { + rci = malloc(sizeof(rtp_channel_info_t)); + rci->call_num = rsi->call_num; + rci->start_time = rsi->start_time; + rci->end_time = rsi->start_time; + rci->selected = FALSE; + rci->frame_index = 0; + rci->drop_by_jitter_buff = 0; + rci->out_of_seq = 0; + rci->max_frame_index = 0; + rci->samples = g_array_new (FALSE, FALSE, sizeof(sample_t)); + rci->check_bt = NULL; + rci->separator = NULL; + rci->draw_area = NULL; + rci->pixmap = NULL; + rci->h_scrollbar_adjustment = NULL; + rci->cursor_pixbuf = NULL; + rci->cursor_prev = 0; + rci->cursor_catch = FALSE; + rci->first_stream = rsi; + rci->num_packets = rsi->num_packets; + g_hash_table_insert(rtp_channels_hash, g_strdup(key_str->str), rci); + } else { + /* Add silence between the two streams if needed */ + silence_frames = (gint32)( ((rsi->start_time - rci->end_time)/1000)*SAMPLE_RATE ); + for (i = 0; i< silence_frames; i++) { + g_array_append_val(rci->samples, silence); + } + rci->num_packets += rsi->num_packets; + } + + /* decode the RTP stream */ + first = TRUE; + rtp_time = 0; + decoded_bytes = 0; + decoded_bytes_prev = 0; + silence_frames = 0; + arrive_time = start_time = 0; + arrive_time_prev = 0; + pack_period = 0; + total_time = 0; + total_time_prev = 0; + seq = 0; + delay = 0; + prev_diff = 0; + mean_delay = 0; + variation = 0; + start_timestamp = 0; + + /* we update the progress bar 100 times */ + progbar_quantum = total_packets/100; + progbar_nextstep = progbar_count; + + status = S_NORMAL; + + rtp_packets_list = g_list_first(rsi->rtp_packets_list); + while (rtp_packets_list) + { + + if (progbar_count >= progbar_nextstep) { + g_assert(total_packets > 0); + + progbar_val = (gfloat) progbar_count / total_packets; + + update_progress_bar(progbar_val); + + progbar_nextstep += progbar_quantum; + } + + + rp = rtp_packets_list->data; + if (first == TRUE) { + start_timestamp = rp->info->info_timestamp; /* defined start_timestmp to avoid overflow in timestamp. TODO: handle the timestamp correctly */ + start_rtp_time = 0; + rtp_time_prev = start_rtp_time; + first = FALSE; + seq = rp->info->info_seq_num - 1; + } + + decoded_bytes = decode_rtp_packet(rp, rci, &out_buff); + if (decoded_bytes == 0) { + seq = rp->info->info_seq_num; + } + + rtp_time = (double)(rp->info->info_timestamp-start_timestamp)/SAMPLE_RATE - start_rtp_time; + arrive_time = (double)rp->arrive_offset/1000 - start_time; + + if (rp->info->info_seq_num != seq+1){ + rci->out_of_seq++; + status = S_WRONG_SEQ; + } + seq = rp->info->info_seq_num; + + diff = arrive_time - rtp_time; + + delay = diff - prev_diff; + prev_diff = diff; + if (delay<0) delay = -delay; + + if (diff<0) diff = -diff; + + total_time = (double)rp->arrive_offset/1000; + + printf("seq = %d arr = %f abs_diff = %f index = %d tim = %f ji=%d jb=%f\n",rp->info->info_seq_num, + total_time, diff, rci->samples->len, ((double)rci->samples->len/8000 - total_time)*1000, 0, + (mean_delay + 4*variation)*1000); + fflush(stdout); + + /* if the jitter buffer was exceded */ + if ( diff*1000 > jitter_buff ) { + printf("Packet drop by jitter buffer exceded\n"); + rci->drop_by_jitter_buff++; + status = S_DROP_BY_JITT; + + /* if there was a silence period (more than two packetization period) resync the source */ + if ( (rtp_time - rtp_time_prev) > pack_period*2 ){ + printf("Resync...\n"); + + silence_frames = (gint32)((arrive_time - arrive_time_prev)*SAMPLE_RATE - decoded_bytes_prev/2); + for (i = 0; i< silence_frames; i++) { + silence.status = status; + g_array_append_val(rci->samples, silence); + + /* only mark the fisrt in the silence that has the previos problem (S_DROP_BY_JITT or S_WRONG_SEQ ) */ + status = S_NORMAL; + } + + decoded_bytes_prev = 0; + start_timestamp = rp->info->info_timestamp; /* defined start_timestmp to avoid overflow in timestamp. TODO: handle the timestamp correctly */ + start_rtp_time = 0; + start_time = (double)rp->arrive_offset/1000; + rtp_time_prev = 0; + } + } else { + /* Add silence if it is necessary */ + silence_frames = (gint32)((rtp_time - rtp_time_prev)*SAMPLE_RATE - decoded_bytes_prev/2); + for (i = 0; i< silence_frames; i++) { + silence.status = status; + g_array_append_val(rci->samples, silence); + + /* only mark the fisrt in the silence that has the previos problem (S_DROP_BY_JITT or S_WRONG_SEQ ) */ + status = S_NORMAL; + } + + status = S_NORMAL; + + /* Add the audio */ + for (i = 0; i< (decoded_bytes/2); i++) { + sample.val = out_buff[i]; + sample.status = status; + g_array_append_val(rci->samples, sample); + } + + rtp_time_prev = rtp_time; + pack_period = (double)(decoded_bytes/2)/SAMPLE_RATE; + decoded_bytes_prev = decoded_bytes; + arrive_time_prev = arrive_time; + + } + + rtp_packets_list = g_list_next (rtp_packets_list); + progbar_count++; + } + rci->max_frame_index = rci->samples->len; + rci->end_time = rci->start_time + ((double)rci->samples->len/SAMPLE_RATE)*1000; + + g_string_free(key_str, TRUE); +} + +/****************************************************************************/ +static gint +h_scrollbar_changed(GtkWidget *widget _U_, gpointer user_data) +{ + rtp_channel_info_t *rci = (rtp_channel_info_t *)user_data; + rci->cursor_catch = TRUE; + return TRUE; +} + +static gboolean draw_cursors(gpointer *data); + +/****************************************************************************/ +static void +stop_channels() +{ + PaError err; + GtkWidget *dialog; + + /* we should never be here if we are already in STOP */ + if(rtp_channels->stop){ + exit(10); + } + + rtp_channels->stop = TRUE; + /* force a draw_cursor to stop it */ + draw_cursors(NULL); + + err = Pa_StopStream(pa_stream); + + if( err != paNoError ) { + dialog = gtk_message_dialog_new ((GtkWindow *) listen_rtp_dlg_w, + GTK_DIALOG_DESTROY_WITH_PARENT, GTK_MESSAGE_ERROR,GTK_BUTTONS_CLOSE, + "Can not Stop Stream in PortAduio Library.\n Error: %s", Pa_GetErrorText( err )); + gtk_dialog_run (GTK_DIALOG (dialog)); + gtk_widget_destroy (dialog); + return; + } + + rtp_channels->start_index[0] = 0; + rtp_channels->start_index[1] = 0; + rtp_channels->end_index[0] = 0; + rtp_channels->end_index[1] = 0; + rtp_channels->max_frame_index = 0; + rtp_channels->frame_index = 0; + rtp_channels->pause = FALSE; + rtp_channels->pause_duration = 0; + rtp_channels->stop = TRUE; + rtp_channels->out_diff_time = 10000; + + if (rtp_channels->rci[0]) rtp_channels->rci[0]->frame_index = 0; + if (rtp_channels->rci[1]) rtp_channels->rci[1]->frame_index = 0; + + /* set the sensitive state of the buttons (decode, play, pause, stop) */ + bt_state(TRUE, TRUE, FALSE, FALSE); + +} + +/****************************************************************************/ +/* Draw a cursor in a channel graph + */ +static void +draw_channel_cursor(rtp_channel_info_t *rci, guint32 start_index) +{ + PaTimestamp index; + int i; + + if (!rci) return; + + index = Pa_StreamTime( pa_stream ) - rtp_channels->pause_duration - rtp_channels->out_diff_time - start_index; + + + /* If we finished playing both channels, then stop them */ + if ( (rtp_channels && (!rtp_channels->stop) && (!rtp_channels->pause)) && (index > rtp_channels->max_frame_index) ) { + stop_channels(); + return; + } + + /* If only this channel finished, then return */ + if (index > rci->max_frame_index) { + return; + } + + /* draw the previous saved pixbuf line */ + if (rci->cursor_pixbuf) { + + gdk_draw_pixbuf(rci->pixmap, NULL, rci->cursor_pixbuf, 0, 0, (int) (rci->cursor_prev/MULT), 0, -1, -1, GDK_RGB_DITHER_NONE, 0 ,0); + + gdk_draw_drawable(rci->draw_area->window, + rci->draw_area->style->fg_gc[GTK_WIDGET_STATE(rci->draw_area)], + rci->pixmap, + (int) (rci->cursor_prev/MULT), 0, + (int) (rci->cursor_prev/MULT), 0, + 1, rci->draw_area->allocation.height-HEIGHT_TIME_LABEL); + + g_object_unref(rci->cursor_pixbuf); + } + + rci->cursor_pixbuf = gdk_pixbuf_get_from_drawable(NULL, rci->pixmap, NULL, (int) (index/MULT), 0, 0, 0, 1, rci->draw_area->allocation.height-HEIGHT_TIME_LABEL); + + gdk_draw_line(rci->pixmap, rci->draw_area->style->black_gc, + (int) (index/MULT), + 0, + (int) (index/MULT), + rci->draw_area->allocation.height-HEIGHT_TIME_LABEL); + + gdk_draw_drawable(rci->draw_area->window, + rci->draw_area->style->fg_gc[GTK_WIDGET_STATE(rci->draw_area)], + rci->pixmap, + (int) (index/MULT), 0, + (int) (index/MULT), 0, + 1, rci->draw_area->allocation.height-HEIGHT_TIME_LABEL); + + /* Disconnect the scroll bar "value" signal to not be called */ + SIGNAL_DISCONNECT_BY_FUNC(rci->h_scrollbar_adjustment, h_scrollbar_changed, rci); + + /* Move the horizontal scroll bar */ +/* if ( (rci->cursor_prev/MULT < (rci->h_scrollbar_adjustment->value+rci->h_scrollbar_adjustment->page_increment)) && + (index/MULT >= (rci->h_scrollbar_adjustment->value+rci->h_scrollbar_adjustment->page_increment)) ){ + for (i=1; i<10; i++) { + rci->h_scrollbar_adjustment->value += rci->h_scrollbar_adjustment->page_size/10; + gtk_adjustment_value_changed(rci->h_scrollbar_adjustment); + } + + } +*/ + if (!rci->cursor_catch) { + if (index/MULT < rci->h_scrollbar_adjustment->page_size/2) { + rci->h_scrollbar_adjustment->value = rci->h_scrollbar_adjustment->lower; + } else if (index/MULT > (rci->h_scrollbar_adjustment->upper - rci->h_scrollbar_adjustment->page_size/2)) { + rci->h_scrollbar_adjustment->value = rci->h_scrollbar_adjustment->upper - rci->h_scrollbar_adjustment->page_size; + } else { + rci->h_scrollbar_adjustment->value = index/MULT - rci->h_scrollbar_adjustment->page_size/2; + } + + gtk_adjustment_value_changed(rci->h_scrollbar_adjustment); + } else if ( (rci->cursor_prev/MULT < (rci->h_scrollbar_adjustment->value+rci->h_scrollbar_adjustment->page_increment)) && + (index/MULT >= (rci->h_scrollbar_adjustment->value+rci->h_scrollbar_adjustment->page_increment)) ){ + rci->cursor_catch = FALSE; + for (i=1; i<10; i++) { + rci->h_scrollbar_adjustment->value = min(rci->h_scrollbar_adjustment->upper-rci->h_scrollbar_adjustment->page_size, rci->h_scrollbar_adjustment->value + (rci->h_scrollbar_adjustment->page_size/20)); + gtk_adjustment_value_changed(rci->h_scrollbar_adjustment); + } + + } + + + /* Connect back the "value" scroll signal */ + SIGNAL_CONNECT(rci->h_scrollbar_adjustment, "value_changed", h_scrollbar_changed, rci); + + +/* if (index/MULT < rci->h_scrollbar_adjustment->page_increment) { + rci->h_scrollbar_adjustment->value = rci->h_scrollbar_adjustment->lower; + } else if (index/MULT > (rci->h_scrollbar_adjustment->upper - rci->h_scrollbar_adjustment->page_size + rci->h_scrollbar_adjustment->page_increment)) { + rci->h_scrollbar_adjustment->value = rci->h_scrollbar_adjustment->upper - rci->h_scrollbar_adjustment->page_size; + } else { + if ( (index/MULT < rci->h_scrollbar_adjustment->value) || (index/MULT > (rci->h_scrollbar_adjustment->value+rci->h_scrollbar_adjustment->page_increment)) ){ + rci->h_scrollbar_adjustment->value = index/MULT; + } + } +*/ + +/* if (index/MULT < rci->h_scrollbar_adjustment->page_size/2) { + rci->h_scrollbar_adjustment->value = rci->h_scrollbar_adjustment->lower; + } else if (index/MULT > (rci->h_scrollbar_adjustment->upper - rci->h_scrollbar_adjustment->page_size/2)) { + rci->h_scrollbar_adjustment->value = rci->h_scrollbar_adjustment->upper - rci->h_scrollbar_adjustment->page_size; + } else { + rci->h_scrollbar_adjustment->value = index/MULT - rci->h_scrollbar_adjustment->page_size/2; + } +*/ +/* gtk_adjustment_value_changed(rci->h_scrollbar_adjustment); +*/ + rci->cursor_prev = index; +} + +/****************************************************************************/ +/* Move and draw the cursor in the graph + */ +static gboolean +draw_cursors(gpointer *data) +{ + static GdkPixbuf* pixbuf = NULL; + static PaTimestamp prev; + + if (!rtp_channels) return FALSE; + + /* Draw and move each of the two channels */ + draw_channel_cursor(rtp_channels->rci[0], rtp_channels->start_index[0]); + draw_channel_cursor(rtp_channels->rci[1], rtp_channels->start_index[1]); + + if ((rtp_channels->stop) || (rtp_channels->pause)) return FALSE; + + return TRUE; +} + +/****************************************************************************/ +void +init_rtp_channels_vals() +{ + rtp_play_channles_t *rpci = rtp_channels; + + /* if we only have one channel to play, we just use the info from that channel */ + if (rpci->rci[0] == NULL) { + rpci->max_frame_index = rpci->rci[1]->max_frame_index; + rpci->start_index[0] = rpci->max_frame_index; + rpci->start_index[1] = 0; + rpci->end_index[0] = rpci->max_frame_index; + rpci->end_index[1] = rpci->max_frame_index; + } else if (rpci->rci[1] == NULL) { + rpci->max_frame_index = rpci->rci[0]->max_frame_index; + rpci->start_index[1] = rpci->max_frame_index; + rpci->start_index[0] = 0; + rpci->end_index[0] = rpci->max_frame_index; + rpci->end_index[1] = rpci->max_frame_index; + + /* if the two channels are to be played, then we need to sync both based on the start/end time of each one */ + } else { + rpci->max_frame_index = (guint32)(SAMPLE_RATE/1000) * (guint32)(max(rpci->rci[0]->end_time, rpci->rci[1]->end_time) - + (guint32)min(rpci->rci[0]->start_time, rpci->rci[1]->start_time)); + + if (rpci->rci[0]->start_time < rpci->rci[1]->start_time) { + rpci->start_index[0] = 0; + rpci->start_index[1] = (guint32)(SAMPLE_RATE/1000) * (guint32)(rpci->rci[1]->start_time - rpci->rci[0]->start_time); + } else { + rpci->start_index[1] = 0; + rpci->start_index[0] = (guint32)(SAMPLE_RATE/1000) * (guint32)(rpci->rci[0]->start_time - rpci->rci[1]->start_time); + } + + if (rpci->rci[0]->end_time < rpci->rci[1]->end_time) { + rpci->end_index[0] = rpci->max_frame_index - ((guint32)(SAMPLE_RATE/1000) * (guint32)(rpci->rci[1]->end_time - rpci->rci[0]->end_time)); + rpci->end_index[1] = rpci->max_frame_index; + } else { + rpci->end_index[1] = rpci->max_frame_index - ((guint32)(SAMPLE_RATE/1000) * (guint32)(rpci->rci[0]->end_time - rpci->rci[1]->end_time)); + rpci->end_index[0] = rpci->max_frame_index; + } + } +} + + +/****************************************************************************/ +/* This routine will be called by the PortAudio engine when audio is needed. + * It may called at interrupt level on some machines so don't do anything + * that could mess up the system like calling malloc() or free(). + */ +static int paCallback( void *inputBuffer, void *outputBuffer, + unsigned long framesPerBuffer, + PaTimestamp outTime, void *userData) +{ + rtp_play_channles_t *rpci = (rtp_play_channles_t*)userData; + SAMPLE *wptr = (SAMPLE*)outputBuffer; + sample_t sample; + unsigned int i; + int finished; + unsigned int framesLeft; + int framesToPlay; + + /* if it is pasued, we keep the stream running but with silence only */ + if (rtp_channels->pause) { + for(i=0; ipause_duration += framesPerBuffer; + return 0; + } + + rpci->out_diff_time = outTime - Pa_StreamTime(pa_stream) ; + + + /* set the values if this is the first time */ + if (rpci->max_frame_index == 0) { + init_rtp_channels_vals(); + + } + + framesLeft = rpci->max_frame_index - rpci->frame_index; + + (void) inputBuffer; /* Prevent unused variable warnings. */ + (void) outTime; + + if( framesLeft < framesPerBuffer ) + { + framesToPlay = framesLeft; + finished = 1; + } + else + { + framesToPlay = framesPerBuffer; + finished = 0; + } + + for( i=0; i<(unsigned int)framesToPlay; i++ ) + { + if (rpci->rci[0] && ( (rpci->frame_index >= rpci->start_index[0]) && (rpci->frame_index <= rpci->end_index[0]) )) { + sample = g_array_index(rpci->rci[0]->samples, sample_t, rpci->rci[0]->frame_index++); + *wptr++ = sample.val; + } else { + *wptr++ = 0; + } + + if (rpci->rci[1] && ( (rpci->frame_index >= rpci->start_index[1]) && (rpci->frame_index <= rpci->end_index[1]) )) { + sample = g_array_index(rpci->rci[1]->samples, sample_t, rpci->rci[1]->frame_index++); + *wptr++ = sample.val; + } else { + *wptr++ = 0; + } + } + for( ; iframe_index += framesToPlay; + + return finished; +} + +/****************************************************************************/ +static void +on_bt_check_clicked(GtkButton *button _U_, gpointer user_data _U_) +{ + rtp_channel_info_t *rci = user_data; + + if (rci->selected) { + if (rtp_channels->rci[0] == rci) { + rtp_channels->rci[0] = NULL; + rtp_channels->channel = 0; + } else { + rtp_channels->rci[1] = NULL; + rtp_channels->channel = 1; + } + } else { + /* if there are already both channels selected, unselect the old one */ + if (rtp_channels->rci[rtp_channels->channel]) { + /* we disconnect the signal temporarly to avoid been called back */ + SIGNAL_DISCONNECT_BY_FUNC(rtp_channels->rci[rtp_channels->channel]->check_bt, on_bt_check_clicked, rtp_channels->rci[rtp_channels->channel]); + gtk_toggle_button_set_active((GtkToggleButton *)rtp_channels->rci[rtp_channels->channel]->check_bt, FALSE); + SIGNAL_CONNECT(rtp_channels->rci[rtp_channels->channel]->check_bt, "clicked", on_bt_check_clicked, rtp_channels->rci[rtp_channels->channel]); + rtp_channels->rci[rtp_channels->channel]->selected = FALSE; + } + + rtp_channels->rci[rtp_channels->channel] = rci; + rtp_channels->channel = !(rtp_channels->channel); + } + + rci->selected = !(rci->selected); + + /* set the sensitive state of the buttons (decode, play, pause, stop) */ + bt_state(TRUE, TRUE, FALSE, FALSE); +} + +/****************************************************************************/ +static void channel_draw(rtp_channel_info_t* rci) +{ + int i,j; + sample_t sample; + SAMPLE min, max; + PangoLayout *small_layout; + guint32 label_width, label_height; + char label_string[MAX_TIME_LABEL]; + double offset; + guint32 progbar_nextstep; + int progbar_quantum; + gfloat progbar_val; + guint status; + GdkGC *gc; + GdkGC *red_gc; + GdkColor red_color = {0, 65535, 0, 0}; + + if (GDK_IS_DRAWABLE(rci->pixmap)) { + /* Clear out old plot */ + gdk_draw_rectangle(rci->pixmap, + rci->bg_gc[1+rci->call_num%MAX_NUM_COL_CONV], + TRUE, + 0, 0, + rci->draw_area->allocation.width, + rci->draw_area->allocation.height); + + small_layout = gtk_widget_create_pango_layout(rci->draw_area, label_string); + pango_layout_set_font_description(small_layout, pango_font_description_from_string("Helvetica,Sans,Bold 7")); + + /* calculated the pixel offset to display integer seconds */ + offset = ((double)rci->start_time/1000 - floor((double)rci->start_time/1000))*SAMPLE_RATE/MULT; + + gdk_draw_line(rci->pixmap, rci->draw_area->style->black_gc, + 0, + rci->draw_area->allocation.height-HEIGHT_TIME_LABEL, + rci->draw_area->allocation.width, + rci->draw_area->allocation.height-HEIGHT_TIME_LABEL); + + /* we update the progress bar 100 times */ + progbar_quantum = (total_frames/MULT)/100; + progbar_nextstep = progbar_count; + + red_gc = gdk_gc_new(rci->draw_area->window); + gdk_gc_set_rgb_fg_color(red_gc, &red_color); + + for (i=0; i< min(rci->draw_area->allocation.width,(gint)(rci->samples->len/MULT)); i++) { + sample.val = 0; + status = S_NORMAL; + max=(SAMPLE)0xFFFF; + min=(SAMPLE)0x7FFF; + + if (progbar_count >= progbar_nextstep) { + g_assert(total_frames > 0); + + progbar_val = (gfloat) progbar_count / (total_frames/MULT); + + update_progress_bar(progbar_val); + + progbar_nextstep += progbar_quantum; + } + + for (j=0; jsamples, sample_t, i*MULT+j); + max = max(max, sample.val); + min = min(min, sample.val); + if (sample.status == S_DROP_BY_JITT) status = S_DROP_BY_JITT; + } + + if (status == S_DROP_BY_JITT) { + gc = red_gc; + } else { + gc = rci->draw_area->style->black_gc; + } + + gdk_draw_line(rci->pixmap, gc, + i, + (gint)(( (0x7FFF+min) * (rci->draw_area->allocation.height-HEIGHT_TIME_LABEL))/0xFFFF), + i, + (gint)(( (0x7FFF+max) * (rci->draw_area->allocation.height-HEIGHT_TIME_LABEL))/0xFFFF)); + + /*draw the time label and grid */ + if ( !((i*MULT)%(SAMPLE_RATE)) ) { + gdk_draw_line(rci->pixmap, rci->draw_area->style->black_gc, + (int) (i - offset), + rci->draw_area->allocation.height-HEIGHT_TIME_LABEL, + (int) (i - offset), + rci->draw_area->allocation.height-HEIGHT_TIME_LABEL+4); + + g_snprintf(label_string, MAX_TIME_LABEL, "%.0f", floor(rci->start_time/1000) + i*MULT/SAMPLE_RATE); + + pango_layout_set_text(small_layout, label_string, -1); + pango_layout_get_pixel_size(small_layout, &label_width, &label_height); + gdk_draw_layout(rci->pixmap, + rci->draw_area->style->black_gc, + (int) (i - offset - label_width/2), + rci->draw_area->allocation.height - label_height, + small_layout); + /* draw the 1/2 sec grid */ + } else if ( !((i*MULT)%(SAMPLE_RATE/2)) ) { + gdk_draw_line(rci->pixmap, rci->draw_area->style->black_gc, + (int) (i - offset), + rci->draw_area->allocation.height-HEIGHT_TIME_LABEL, + (int) (i - offset), + rci->draw_area->allocation.height-HEIGHT_TIME_LABEL+2); + + } + + progbar_count++; + } + } + +} +/****************************************************************************/ +static gint expose_event_channels(GtkWidget *widget, GdkEventExpose *event) +{ + rtp_channel_info_t *rci; + + rci=(rtp_channel_info_t *)OBJECT_GET_DATA(widget, "rtp_channel_info_t"); + if(!rci){ + exit(10); + } + + if (GDK_IS_DRAWABLE(widget->window)) + gdk_draw_drawable(widget->window, + widget->style->fg_gc[GTK_WIDGET_STATE(widget)], + rci->pixmap, + event->area.x, event->area.y, + event->area.x, event->area.y, + event->area.width, event->area.height); + + return FALSE; +} + +/****************************************************************************/ +static gint +configure_event_channels(GtkWidget *widget, GdkEventConfigure *event _U_) +{ + rtp_channel_info_t *rci; + int i; + + /* the first calor is blue to highlight the selected item + * the other collors are the same as in the Voip Graph analysys + * to match the same calls + */ + static GdkColor col[MAX_NUM_COL_CONV+1] = { + {0, 0x00FF, 0x00FF, 0xFFFF}, + {0, 0x33FF, 0xFFFF, 0x33FF}, + {0, 0x00FF, 0xCCFF, 0xCCFF}, + {0, 0x66FF, 0xFFFF, 0xFFFF}, + {0, 0x99FF, 0x66FF, 0xFFFF}, + {0, 0xFFFF, 0xFFFF, 0x33FF}, + {0, 0xCCFF, 0x99FF, 0xFFFF}, + {0, 0xCCFF, 0xFFFF, 0x33FF}, + {0, 0xFFFF, 0xCCFF, 0xCCFF}, + {0, 0xFFFF, 0x99FF, 0x66FF}, + {0, 0xFFFF, 0xFFFF, 0x99FF} + }; + + rci=(rtp_channel_info_t *)OBJECT_GET_DATA(widget, "rtp_channel_info_t"); + if(!rci){ + exit(10); + } + + if(rci->pixmap){ + g_object_unref(rci->pixmap); + rci->pixmap=NULL; + } + + rci->pixmap = gdk_pixmap_new(widget->window, + widget->allocation.width, + widget->allocation.height, + -1); + + if ( GDK_IS_DRAWABLE(rci->pixmap) ) + gdk_draw_rectangle(rci->pixmap, + widget->style->white_gc, + TRUE, + 0, 0, + widget->allocation.width, + widget->allocation.height); + + /* create gcs for the background color of each channel */ + for (i=0; ibg_gc[i]=gdk_gc_new(rci->pixmap); + gdk_gc_set_rgb_fg_color(rci->bg_gc[i], &col[i]); + } + + channel_draw(rci); + + return TRUE; +} + +/****************************************************************************/ +static gint +button_press_event_channel(GtkWidget *widget, GdkEventButton *event _U_) +{ + rtp_channel_info_t *rci; + int this_channel; + guint32 prev_index; + + rci=(rtp_channel_info_t *)OBJECT_GET_DATA(widget, "rtp_channel_info_t"); + if(!rci){ + exit(10); + } + + if (!rci->selected) { + + /* only select a new channels if we are in STOP */ + if (!rtp_channels->stop) return 0; + + /* if there are already both channels selected, unselect the old one */ + if (rtp_channels->rci[rtp_channels->channel]) { + /* we disconnect the signal temporarly to avoid been called back */ + SIGNAL_DISCONNECT_BY_FUNC(rtp_channels->rci[rtp_channels->channel]->check_bt, on_bt_check_clicked, rtp_channels->rci[rtp_channels->channel]); + gtk_toggle_button_set_active((GtkToggleButton *) rtp_channels->rci[rtp_channels->channel]->check_bt, FALSE); + SIGNAL_CONNECT(rtp_channels->rci[rtp_channels->channel]->check_bt, "clicked", on_bt_check_clicked, rtp_channels->rci[rtp_channels->channel]); + rtp_channels->rci[rtp_channels->channel]->selected = FALSE; + } + + /* we disconnect the signal temporarly to avoid been called back */ + SIGNAL_DISCONNECT_BY_FUNC(rci->check_bt, on_bt_check_clicked, rci); + gtk_toggle_button_set_active((GtkToggleButton *) rci->check_bt, TRUE); + SIGNAL_CONNECT(rci->check_bt, "clicked", on_bt_check_clicked, rci); + + rtp_channels->rci[rtp_channels->channel] = rci; + rtp_channels->channel = !(rtp_channels->channel); + rci->selected = TRUE; + + /* set the sensitive state of the buttons (decode, play, pause, stop) */ + bt_state(TRUE, TRUE, FALSE, FALSE); + } + + if (rci == rtp_channels->rci[0]) { + this_channel = 0; + } else { + this_channel = 1; + } + + rci->frame_index = (unsigned int) (event->x * MULT); + + prev_index = rtp_channels->frame_index; + rtp_channels->frame_index = rtp_channels->start_index[this_channel] + rci->frame_index; + rtp_channels->pause_duration += prev_index - rtp_channels->frame_index; + + + + /* change the index in the other channel if selected, according with the index position */ + if (rtp_channels->rci[!this_channel]) { + init_rtp_channels_vals(); + + if (rtp_channels->frame_index < rtp_channels->start_index[!this_channel]) { + rtp_channels->rci[!this_channel]->frame_index = 0; + } else if (rtp_channels->frame_index > rtp_channels->end_index[!this_channel]) { + rtp_channels->rci[!this_channel]->frame_index = rtp_channels->rci[!this_channel]->max_frame_index; + } else { + rtp_channels->rci[!this_channel]->frame_index = rtp_channels->frame_index - rtp_channels->start_index[!this_channel]; + } + } else { + init_rtp_channels_vals(); + } + + rtp_channels->out_diff_time = 0; + + rci->cursor_catch = TRUE; + + /* redraw the cusrsor */ + draw_cursors(NULL); + + return TRUE; +} + +/****************************************************************************/ +static void +/*add_channel_to_window(gchar *key _U_ , rtp_channel_info_t *rci, gpointer ptr _U_ ) */ +add_channel_to_window(gchar *key _U_ , rtp_channel_info_t *rci, guint *counter _U_ ) +{ + GString *label = NULL; + GtkWidget *viewport; + + + /* create the channel draw area */ + rci->draw_area=gtk_drawing_area_new(); + + rci->scroll_window=gtk_scrolled_window_new(NULL, NULL); + + gtk_scrolled_window_set_policy (GTK_SCROLLED_WINDOW (rci->scroll_window), GTK_POLICY_ALWAYS, GTK_POLICY_NEVER); + rci->h_scrollbar_adjustment = gtk_scrolled_window_get_hadjustment(GTK_SCROLLED_WINDOW(rci->scroll_window)); + + + gtk_widget_set_size_request(rci->draw_area, (gint)(rci->samples->len/MULT), CHANNEL_HEIGHT); + + + viewport = gtk_viewport_new(rci->h_scrollbar_adjustment, gtk_scrolled_window_get_vadjustment(GTK_SCROLLED_WINDOW(rci->scroll_window))); + gtk_container_add(GTK_CONTAINER(viewport), rci->draw_area); + gtk_container_add(GTK_CONTAINER(rci->scroll_window), viewport); + gtk_viewport_set_shadow_type(GTK_VIEWPORT(viewport), GTK_SHADOW_NONE); + OBJECT_SET_DATA(rci->draw_area, "rtp_channel_info_t", rci); + gtk_widget_add_events (rci->draw_area, GDK_BUTTON_PRESS_MASK); + GTK_WIDGET_SET_FLAGS(rci->draw_area, GTK_CAN_FOCUS); + gtk_widget_grab_focus(rci->draw_area); + + gtk_box_pack_start(GTK_BOX (channels_vb), rci->scroll_window, FALSE, FALSE, 0); + + /* signals needed to handle backing pixmap */ + SIGNAL_CONNECT(rci->draw_area, "expose_event", expose_event_channels, NULL); + SIGNAL_CONNECT(rci->draw_area, "configure_event", configure_event_channels, rci); + gtk_widget_add_events (rci->draw_area, GDK_BUTTON_PRESS_MASK); + SIGNAL_CONNECT(rci->draw_area, "button_press_event", button_press_event_channel, rci); + SIGNAL_CONNECT(rci->h_scrollbar_adjustment, "value_changed", h_scrollbar_changed, rci); + + + label = g_string_new(""); + g_string_printf(label, "From %s:%d to %s:%d Duration:%.2f Drop by Jitter Buff:%d(%.1f%%) Out of Seq: %d(%.1f%%)", get_addr_name(&(rci->first_stream->src_addr)), + rci->first_stream->src_port, get_addr_name(&(rci->first_stream->dest_addr)), rci->first_stream->dest_port, + (double)rci->samples->len/SAMPLE_RATE, rci->drop_by_jitter_buff, (double)rci->drop_by_jitter_buff * 100 / (double)rci->num_packets + , rci->out_of_seq, (double)rci->out_of_seq * 100 / (double)rci->num_packets); + + rci->check_bt = gtk_check_button_new_with_label(label->str); + gtk_box_pack_start(GTK_BOX (channels_vb), rci->check_bt, FALSE, FALSE, 1); + + /* Create the Separator if it is not the last one */ + (*counter)++; + if (*counter < g_hash_table_size(rtp_channels_hash)) { + rci->separator = gtk_hseparator_new(); + gtk_box_pack_start(GTK_BOX (channels_vb), rci->separator, FALSE, FALSE, 5); + } + + SIGNAL_CONNECT(rci->check_bt, "clicked", on_bt_check_clicked, rci); + + g_string_free(label, TRUE); +} + +/****************************************************************************/ +static void +count_channel_frames(gchar *key _U_ , rtp_channel_info_t *rci, gpointer ptr _U_ ) +{ + total_frames += rci->samples->len; +} + +/****************************************************************************/ +static void +play_channels() +{ + PaError err; + GtkWidget *dialog; + + /* we should never be here if we are in PLAY and !PAUSE */ + if(!rtp_channels->stop && !rtp_channels->pause){ + exit(10); + } + + /* if we are in PAUSE change the sate */ + if (rtp_channels->pause) { + rtp_channels->pause = FALSE; + /* set the sensitive state of the buttons (decode, play, pause, stop) */ + bt_state(FALSE, FALSE, TRUE, TRUE); + + /* if not PAUSE, then start to PLAY */ + } else { + err = Pa_OpenStream( + &pa_stream, + paNoDevice, /* default input device */ + 0, /* no input */ + PA_SAMPLE_TYPE, /* 16 bit Integer input */ + NULL, + Pa_GetDefaultOutputDeviceID(), + NUM_CHANNELS, /* Stereo output */ + PA_SAMPLE_TYPE, /* 16 bit Integer output */ + NULL, + SAMPLE_RATE, + FRAMES_PER_BUFFER, + 0, /* number of buffers, if zero then use default minimum */ + paClipOff, /* we won't output out of range samples so don't bother clipping them */ + paCallback, + rtp_channels ); + + if( err != paNoError ) { + dialog = gtk_message_dialog_new ((GtkWindow *) listen_rtp_dlg_w, + GTK_DIALOG_DESTROY_WITH_PARENT, GTK_MESSAGE_ERROR,GTK_BUTTONS_CLOSE, + "Can not Open Stream in PortAduio Library.\n Error: %s", Pa_GetErrorText( err )); + gtk_dialog_run (GTK_DIALOG (dialog)); + gtk_widget_destroy (dialog); + return; + } + + err = Pa_StartStream( pa_stream ); + if( err != paNoError ) { + dialog = gtk_message_dialog_new ((GtkWindow *) listen_rtp_dlg_w, + GTK_DIALOG_DESTROY_WITH_PARENT, GTK_MESSAGE_ERROR,GTK_BUTTONS_CLOSE, + "Can not Start Stream in PortAduio Library.\n Error: %s", Pa_GetErrorText( err )); + gtk_dialog_run (GTK_DIALOG (dialog)); + gtk_widget_destroy (dialog); + return; + } + + rtp_channels->stop = FALSE; + + /* set the sensitive state of the buttons (decode, play, pause, stop) */ + bt_state(FALSE, FALSE, TRUE, TRUE); + } + + /* Draw the cursor in the graph */ + g_timeout_add_full(G_PRIORITY_DEFAULT_IDLE, MULT*1000/SAMPLE_RATE, draw_cursors, NULL, NULL); + +} + +/****************************************************************************/ +static void +pause_channels() +{ + rtp_channels->pause = !(rtp_channels->pause); + + /* reactivate the cusrosr display if no in pause */ + if (!rtp_channels->pause) { + /* Draw the cursor in the graph */ + g_timeout_add_full(G_PRIORITY_DEFAULT_IDLE, MULT*1000/SAMPLE_RATE, draw_cursors, NULL, NULL); + } + + /* set the sensitive state of the buttons (decode, play, pause, stop) */ + bt_state(FALSE, TRUE, FALSE, TRUE); +} + +/****************************************************************************/ +static void +reset_rtp_channels() +{ + rtp_channels->channel = 0; + rtp_channels->rci[0] = NULL; + rtp_channels->rci[1] = NULL; + rtp_channels->start_index[0] = 0; + rtp_channels->start_index[1] = 0; + rtp_channels->end_index[0] = 0; + rtp_channels->end_index[1] = 0; + rtp_channels->max_frame_index = 0; + rtp_channels->frame_index = 0; + rtp_channels->pause = FALSE; + rtp_channels->pause_duration = 0; + rtp_channels->stop = TRUE; + rtp_channels->out_diff_time = 10000; +} + +/****************************************************************************/ +static void +remove_channel_to_window(gchar *key _U_ , rtp_channel_info_t *rci, gpointer ptr _U_ ) +{ + g_object_unref(rci->pixmap); + gtk_widget_destroy(rci->draw_area); + gtk_widget_destroy(rci->scroll_window); + gtk_widget_destroy(rci->check_bt); + gtk_widget_destroy(rci->separator); +} + +/****************************************************************************/ +static void +reset_channels() +{ + + /* Remove the channels from the main window if there are there */ + g_hash_table_foreach( rtp_channels_hash, (GHFunc)remove_channel_to_window, NULL); + + + /* destroy the rtp channels hash table */ + if (rtp_channels_hash) { + g_hash_table_destroy(rtp_channels_hash); + rtp_channels_hash = NULL; + } + + if (rtp_channels) { + reset_rtp_channels(); + } +} + +/****************************************************************************/ +static void +reset_listen_rtp() +{ + /* Destroy the rtp channels */ + reset_channels(); + + /* destroy the rtp streams hash table */ + if (rtp_streams_hash) { + g_hash_table_destroy(rtp_streams_hash); + rtp_streams_hash = NULL; + } + + /* destroy the rtp streams list */ + if (rtp_streams_list) { + g_list_free (rtp_streams_list); + rtp_streams_list = NULL; + } + +} + +/****************************************************************************/ +static void +decode_streams() +{ + guint statusbar_context; + guint counter; + + /* set the sensitive state of the buttons (decode, play, pause, stop) */ + bt_state(FALSE, FALSE, FALSE, FALSE); + + reset_channels(); + + progress_bar = gtk_progress_bar_new(); + WIDGET_SET_SIZE(progress_bar, 100, -1); + gtk_box_pack_start(GTK_BOX (stat_hbox), progress_bar, FALSE, FALSE, 2); + gtk_widget_show(progress_bar); + statusbar_context = gtk_statusbar_get_context_id((GtkStatusbar *) info_bar, "main"); + gtk_statusbar_push((GtkStatusbar *) info_bar, statusbar_context, " Decoding RTP packets..."); + + gtk_statusbar_set_has_resize_grip(GTK_STATUSBAR(info_bar), FALSE); + + /* reset the number of packet to be decoded, this is used for the progress bar */ + total_packets = 0; + /* reset the Progress Bar count */ + progbar_count = 0; + + /* Mark the RTP streams to be played using the selected VoipCalls*/ + g_hash_table_foreach( rtp_streams_hash, (GHFunc)mark_rtp_stream_to_play, NULL); + + /* Decode the RTP streams and add them to the RTP channels to be played */ + g_list_foreach( rtp_streams_list, (GFunc)decode_rtp_stream, NULL); + + /* reset the number of frames to be displayed, this is used for the progress bar */ + total_frames = 0; + /* Count the frames in all the RTP channels */ + g_hash_table_foreach( rtp_channels_hash, (GHFunc)count_channel_frames, NULL); + + /* reset the Progress Bar count again for the progress of creating the channels view */ + progbar_count = 0; + gtk_statusbar_pop((GtkStatusbar *) info_bar, statusbar_context); + gtk_statusbar_push((GtkStatusbar *) info_bar, statusbar_context, " Creating channels view..."); + + /* Display the RTP channels in the window */ + counter = 0; + g_hash_table_foreach( rtp_channels_hash, (GHFunc)add_channel_to_window, &counter); + + /* Resize the main scroll window to display no more than 5 channels, otherwise the scroll bar need to be used */ + WIDGET_SET_SIZE(main_scrolled_window, CHANNEL_WIDTH, + min(g_hash_table_size(rtp_channels_hash), 5) * (CHANNEL_HEIGHT+60)); + + gtk_widget_show_all(main_scrolled_window); + + gtk_widget_destroy(progress_bar); + gtk_statusbar_set_has_resize_grip(GTK_STATUSBAR(info_bar), TRUE); + gtk_statusbar_pop((GtkStatusbar *) info_bar, statusbar_context); + + /* blank the status label */ + gtk_statusbar_pop((GtkStatusbar *) info_bar, statusbar_context); + + /* set the sensitive state of the buttons (decode, play, pause, stop) */ + bt_state(TRUE, FALSE, FALSE, FALSE); + + /* get the static jitter buffer from the spinner gui */ + new_jitter_buff = (int) gtk_spin_button_get_value((GtkSpinButton * )jitter_spinner); + +} + +/****************************************************************************/ +static void +on_bt_decode_clicked(GtkButton *button _U_, gpointer user_data _U_) +{ + decode_streams(); +} + +/****************************************************************************/ +static void +on_bt_play_clicked(GtkButton *button _U_, gpointer user_data _U_) +{ + play_channels(); +} + +/****************************************************************************/ +static void +on_bt_pause_clicked(GtkButton *button _U_, gpointer user_data _U_) +{ + pause_channels(); +} + +/****************************************************************************/ +static void +on_bt_stop_clicked(GtkButton *button _U_, gpointer user_data _U_) +{ + stop_channels(); +} + +/****************************************************************************/ +static void +listen_rtp_on_destroy(GtkObject *object _U_, gpointer user_data _U_) +{ + /* Stop the channels if necesary */ + if(rtp_channels && (!rtp_channels->stop)){ + stop_channels(); + } + + /* Destroy the rtp channels */ + reset_channels(); + + g_free(rtp_channels); + rtp_channels = NULL; + + initialized = FALSE; + + gtk_widget_destroy(listen_rtp_dlg_w); + main_scrolled_window = NULL; + listen_rtp_dlg_w = NULL; +} + +/****************************************************************************/ +static void +jitter_spinner_value_changed (GtkSpinButton *spinner, gpointer user_data _U_) +{ + /* set the sensitive state of the buttons (decode, play, pause, stop) */ + bt_state(TRUE, TRUE, FALSE, FALSE); +} + +/****************************************************************************/ +static void listen_rtp_dlg_create() +{ + GtkWidget *main_vb; + GtkWidget *hbuttonbox; + GtkWidget *h_jitter_buttons_box; + GtkWidget *bt_close; + GtkAdjustment *jitter_spinner_adj; + GtkWidget *label; + + GtkTooltips *tooltips = gtk_tooltips_new(); + + listen_rtp_dlg_w=gtk_window_new(GTK_WINDOW_TOPLEVEL); + + gtk_window_set_title(GTK_WINDOW(listen_rtp_dlg_w), "Wireshark: Listen RTP"); + gtk_window_set_position(GTK_WINDOW(listen_rtp_dlg_w), GTK_WIN_POS_NONE); + + gtk_window_set_default_size(GTK_WINDOW(listen_rtp_dlg_w), 400, 50); + + main_vb = gtk_vbox_new (FALSE, 0); + gtk_container_add(GTK_CONTAINER(listen_rtp_dlg_w), main_vb); + gtk_container_set_border_width (GTK_CONTAINER (main_vb), 2); + + main_scrolled_window=gtk_scrolled_window_new(NULL, NULL); + gtk_container_set_border_width (GTK_CONTAINER (main_scrolled_window), 4); + WIDGET_SET_SIZE(main_scrolled_window, CHANNEL_WIDTH, 0); + + gtk_scrolled_window_set_policy (GTK_SCROLLED_WINDOW (main_scrolled_window), GTK_POLICY_NEVER, GTK_POLICY_AUTOMATIC); + gtk_container_add(GTK_CONTAINER(main_vb), main_scrolled_window); + + channels_vb = gtk_vbox_new (FALSE, 0); + gtk_container_set_border_width (GTK_CONTAINER (channels_vb), 2); + gtk_scrolled_window_add_with_viewport((GtkScrolledWindow *) main_scrolled_window, channels_vb); + + h_jitter_buttons_box = gtk_hbox_new (FALSE, 0); + gtk_container_set_border_width (GTK_CONTAINER (h_jitter_buttons_box), 10); + gtk_box_pack_start (GTK_BOX(main_vb), h_jitter_buttons_box, FALSE, FALSE, 0); + label = gtk_label_new("Jitter buffer [ms] "); + gtk_box_pack_start(GTK_BOX(h_jitter_buttons_box), label, FALSE, FALSE, 0); + + jitter_spinner_adj = (GtkAdjustment *) gtk_adjustment_new (50, 0, 500, 5, 10, 10); + jitter_spinner = gtk_spin_button_new (jitter_spinner_adj, 5, 0); + gtk_box_pack_start(GTK_BOX(h_jitter_buttons_box), jitter_spinner, FALSE, FALSE, 0); + gtk_tooltips_set_tip (tooltips, jitter_spinner, "The simulated jitter buffer in [ms]", NULL); + SIGNAL_CONNECT(GTK_OBJECT (jitter_spinner_adj), "value_changed", (GtkSignalFunc) jitter_spinner_value_changed, NULL); + + /* button row */ + hbuttonbox = gtk_hbutton_box_new (); + gtk_box_pack_start (GTK_BOX (h_jitter_buttons_box), hbuttonbox, TRUE, TRUE, 0); + gtk_button_box_set_layout (GTK_BUTTON_BOX (hbuttonbox), GTK_BUTTONBOX_SPREAD); + gtk_button_box_set_spacing (GTK_BUTTON_BOX (hbuttonbox), 30); + + bt_decode = gtk_button_new_with_label("Decode"); + gtk_container_add(GTK_CONTAINER(hbuttonbox), bt_decode); + SIGNAL_CONNECT(bt_decode, "clicked", on_bt_decode_clicked, NULL); + gtk_tooltips_set_tip (tooltips, bt_decode, "Decode the RTP stream(s)", NULL); + + bt_play = gtk_button_new_with_label("Play"); + gtk_container_add(GTK_CONTAINER(hbuttonbox), bt_play); + SIGNAL_CONNECT(bt_play, "clicked", on_bt_play_clicked, NULL); + gtk_tooltips_set_tip (tooltips, bt_play, "Play the RTP channel(s)", NULL); + + bt_pause = gtk_button_new_with_label("Pause"); + gtk_container_add(GTK_CONTAINER(hbuttonbox), bt_pause); + SIGNAL_CONNECT(bt_pause, "clicked", on_bt_pause_clicked, NULL); + gtk_tooltips_set_tip (tooltips, bt_pause, "Pause the RTP channel(s)", NULL); + + bt_stop = gtk_button_new_with_label("Stop"); + gtk_container_add(GTK_CONTAINER(hbuttonbox), bt_stop); + SIGNAL_CONNECT(bt_stop, "clicked", on_bt_stop_clicked, NULL); + gtk_tooltips_set_tip (tooltips, bt_stop, "Stop the RTP channel(s)", NULL); + + bt_close = BUTTON_NEW_FROM_STOCK(GTK_STOCK_CLOSE); + gtk_container_add (GTK_CONTAINER (hbuttonbox), bt_close); + GTK_WIDGET_SET_FLAGS(bt_close, GTK_CAN_DEFAULT); + gtk_tooltips_set_tip (tooltips, bt_close, "Close this dialog", NULL); + + SIGNAL_CONNECT(bt_close, "clicked", listen_rtp_on_destroy, NULL); + SIGNAL_CONNECT(listen_rtp_dlg_w, "destroy", listen_rtp_on_destroy, NULL); + + /* button row */ + hbuttonbox = gtk_hbutton_box_new (); + + /* Filter/status hbox */ + stat_hbox = gtk_hbox_new(FALSE, 1); + gtk_container_border_width(GTK_CONTAINER(stat_hbox), 0); + + /* statusbar */ + info_bar = gtk_statusbar_new(); + gtk_statusbar_set_has_resize_grip(GTK_STATUSBAR(info_bar), TRUE); + + gtk_box_pack_start(GTK_BOX(stat_hbox), info_bar, TRUE, TRUE, 0); + + /* statusbar hbox */ + gtk_box_pack_start(GTK_BOX(main_vb), stat_hbox, FALSE, TRUE, 0); + + /* set the sensitive state of the buttons (decode, play, pause, stop) */ + bt_state(TRUE, FALSE, FALSE, FALSE); + + gtk_widget_show_all(listen_rtp_dlg_w); + + /* Force gtk to redraw the window before starting decoding the packet */ + while (g_main_context_iteration(NULL, FALSE)); +} + +/****************************************************************************/ +static void +listen_rtp_init(voip_calls_tapinfo_t *voip_calls_tap) +{ + PaError err; + GtkWidget *dialog; + + if (initialized) return; + initialized = TRUE; + + voip_calls = voip_calls_tap; + err = Pa_Initialize(); + if( err != paNoError ) { + dialog = gtk_message_dialog_new ((GtkWindow *) listen_rtp_dlg_w, + GTK_DIALOG_DESTROY_WITH_PARENT, GTK_MESSAGE_ERROR,GTK_BUTTONS_CLOSE, + "Can not Initialize the PortAduio Library.\n Error: %s", Pa_GetErrorText( err )); + gtk_dialog_run (GTK_DIALOG (dialog)); + gtk_widget_destroy (dialog); + initialized = FALSE; + return; + } + + new_jitter_buff = -1; + +#ifdef HAVE_G729_G723 + /* Initialize the G729 and G723 decoders */ + initG723(); + initG729(); +#endif + + if (!rtp_channels) { + rtp_channels = g_malloc(sizeof(rtp_play_channles_t)); + } + + reset_rtp_channels(); + + /* create the dialog window */ + listen_rtp_dlg_create(); + +} + +/****************************************************************************/ +/* this function will be called using a pseudo dissector, so based on the parameters passed will do: + * voip_calls_tap=rtp=pinfo=NULL -> Reset and close the window if open (called when retapping or close) + * voip_calls_tap != NULL -> listen_rtp_dlg_create (this is called when the user click "listen" in the VoIP calls) + * rtp and pinfo != NULL -> every time there is an RTP packet tap by the VoipCalls + */ +static void listen_rtp_main(voip_calls_tapinfo_t *voip_calls_tap,struct _rtp_info *rtp_info, packet_info *pinfo) +{ + /* Init the pa and create the Dialog */ + if (voip_calls_tap) { + listen_rtp_init(voip_calls_tap); + return; + } + + /* This is a RTP packet to be added */ + if( rtp_info && pinfo) { + add_rtp_packet(rtp_info, pinfo); + return; + } + + /* All three parameters are NULL, so it is a Reset */ + reset_listen_rtp(); +} + +/****************************************************************************/ +extern void proto_reg_handoff_listen_rtp(void) +{ + GString* tap_error = NULL; +} + +/****************************************************************************/ +void proto_register_listen_rtp(void) { + + proto_listen_rtp = proto_register_protocol("Listen RTP", "ListenRTP", "listenrtp"); + register_dissector("listenrtp", listen_rtp_main, proto_listen_rtp); +} + diff --git a/plugins/listen_rtp/main_listen_rtp.h b/plugins/listen_rtp/main_listen_rtp.h new file mode 100644 index 0000000000..2a290afdee --- /dev/null +++ b/plugins/listen_rtp/main_listen_rtp.h @@ -0,0 +1,29 @@ +/* pinfo_stats_tree.h +* Stats tree for ethernet frames +* +* (c) 2006, Alejandro Vaquero +* +* $Id$ +* +* Ethereal - Network traffic analyzer +* By Gerald Combs +* Copyright 1998 Gerald Combs +* +* This program is free software; you can redistribute it and/or +* modify it under the terms of the GNU General Public License +* as published by the Free Software Foundation; either version 2 +* of the License, or (at your option) any later version. +* +* This program 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 General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. +*/ + +extern void proto_register_listen_rtp(void); +extern void proto_reg_handoff_listen_rtp(void); +