Start moving RTP decoding routines to the ui directory.

Move decode_rtp_packet to ui/rtp_media.[ch].

Change-Id: Ib138781c37ac17b807bf75f9d772351aadf72071
Reviewed-on: https://code.wireshark.org/review/10575
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>

5 files changed, 234 insertions, 89 deletions

diff --git a/ui/CMakeLists.txt b/ui/CMakeLists.txt
index 6bf323733a..3a36920839 100644
--- a/ui/CMakeLists.txt
+++ b/ui/CMakeLists.txt
@@ -44,6 +44,7 @@ set(COMMON_UI_SRC
 	profile.c
 	proto_hier_stats.c
 	recent.c
+	rtp_media.c
 	rtp_stream.c
 	service_response_time.c
 	software_update.c
@@ -63,6 +64,7 @@ set(COMMON_UI_SRC
 
 # Enables visibility in IDEs
 file(GLOB EXTRA_UI_HEADERS
+	rtp_media.h
 	rtp_stream.h
 	tap-iax2-analysis.h
 	tap-rtp-analysis.h
diff --git a/ui/Makefile.common b/ui/Makefile.common
index 81bd16ffa0..15ce1010c1 100644
--- a/ui/Makefile.common
+++ b/ui/Makefile.common
@@ -65,6 +65,7 @@ WIRESHARK_UI_SRC = \
 	profile.c		\
 	proto_hier_stats.c	\
 	recent.c		\
+	rtp_media.c		\
 	rtp_stream.c		\
 	service_response_time.c	\
 	software_update.c	\
@@ -107,6 +108,7 @@ noinst_HEADERS = \
 	proto_hier_stats.h	\
 	recent.h		\
 	recent_utils.h		\
+	rtp_media.h		\
 	rtp_stream.h		\
 	service_response_time.h	\
 	simple_dialog.h		\
diff --git a/ui/gtk/rtp_player.c b/ui/gtk/rtp_player.c
index 50bbf70fbe..2774544926 100644
--- a/ui/gtk/rtp_player.c
+++ b/ui/gtk/rtp_player.c
@@ -66,10 +66,9 @@
 #include <epan/prefs.h>
 #include <wsutil/report_err.h>
 
-#include <codecs/codecs.h>
-
 #include "globals.h"
 
+#include "ui/rtp_media.h"
 #include "ui/rtp_stream.h"
 #include "ui/simple_dialog.h"
 #include "ui/voip_calls.h"
@@ -127,7 +126,6 @@ static unsigned channels = 1;
 static unsigned output_channels = 2;
 
 #define PA_SAMPLE_TYPE  paInt16
-typedef gint16 SAMPLE;
 #define SAMPLE_SILENCE  (0)
 #define FRAMES_PER_BUFFER  (512)
 
@@ -190,13 +188,6 @@ typedef struct _rtp_channel_info {
 	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 beginning of the stream in ms */
-	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 */
@@ -219,11 +210,6 @@ typedef struct _rtp_play_channles {
 /* The two RTP channels to play */
 static rtp_play_channels_t *rtp_channels = NULL;
 
-typedef struct _rtp_decoder_t {
-	codec_handle_t handle;
-	void *context;
-} rtp_decoder_t;
-
 
 typedef struct _data_info {
 	int current_channel;
@@ -276,17 +262,6 @@ rtp_stream_value_destroy(gpointer rsi_arg)
 
 /****************************************************************************/
 static void
-rtp_decoder_value_destroy(gpointer dec_arg)
-{
-	rtp_decoder_t *dec = (rtp_decoder_t *)dec_arg;
-
-	if (dec->handle)
-		codec_release(dec->handle, dec->context);
-	g_free(dec_arg);
-}
-
-/****************************************************************************/
-static void
 set_sensitive_check_bt(gchar *key _U_ , rtp_channel_info_t *rci, guint *stop_p )
 {
 	gtk_widget_set_sensitive(rci->check_bt, !(*stop_p));
@@ -379,14 +354,14 @@ add_rtp_packet(const struct _rtp_info *rtp_info, packet_info *pinfo)
 	}
 
 	/* increment the number of packets in this stream, this is used for the progress bar and statistics */
-    stream_info->packet_count++;
+	stream_info->packet_count++;
 
 	/* Add the RTP packet to the list */
-    new_rtp_packet = g_new0(rtp_packet_t, 1);
+	new_rtp_packet = g_new0(rtp_packet_t, 1);
 	new_rtp_packet->info = (struct _rtp_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->rel_ts) - nstime_to_msec(&stream_info->start_rel_time);
+	new_rtp_packet->arrive_offset = nstime_to_msec(&pinfo->rel_ts) - nstime_to_msec(&stream_info->start_rel_time);
 	/* copy the RTP payload to the rtp_packet to be decoded later */
 	if (rtp_info->info_all_data_present && (rtp_info->info_payload_len != 0)) {
 		new_rtp_packet->payload_data = (guint8 *)g_malloc(rtp_info->info_payload_len);
@@ -395,7 +370,7 @@ add_rtp_packet(const struct _rtp_info *rtp_info, packet_info *pinfo)
 		new_rtp_packet->payload_data = NULL;
 	}
 
-    stream_info->rtp_packet_list = g_list_append(stream_info->rtp_packet_list, new_rtp_packet);
+	stream_info->rtp_packet_list = g_list_append(stream_info->rtp_packet_list, new_rtp_packet);
 
 	g_string_free(key_str, TRUE);
 }
@@ -415,7 +390,7 @@ mark_rtp_stream_to_play(gchar *key _U_ , rtp_stream_info_t *rsi, gpointer ptr _U
 	/* Reset the "to be play" value because the user can close and reopen the RTP Player window
 	 * and the streams are not reset in that case
 	 */
-    rsi->decode = FALSE;
+	rsi->decode = FALSE;
 
 	/* and associate the RTP stream with a call using the first RTP packet in the stream */
 	graph_list = g_queue_peek_nth_link(voip_calls->graph_analysis->items, 0);
@@ -454,62 +429,6 @@ mark_all_rtp_stream_to_decode(gchar *key _U_ , rtp_stream_info_t *rsi, gpointer
 }
 
 /****************************************************************************/
-/* Decode a RTP packet
- * Return the number of decoded bytes
- */
-static size_t
-decode_rtp_packet(rtp_packet_t *rp, SAMPLE **out_buff, GHashTable *decoders_hash)
-{
-	unsigned int  payload_type;
-	const gchar *p;
-	rtp_decoder_t *decoder;
-	SAMPLE *tmp_buff = NULL;
-	size_t tmp_buff_len;
-	size_t decoded_bytes = 0;
-
-	if ((rp->payload_data == NULL) || (rp->info->info_payload_len == 0) ) {
-		return 0;
-	}
-
-	payload_type = rp->info->info_payload_type;
-
-	/* Look for registered codecs */
-	decoder = (rtp_decoder_t *)g_hash_table_lookup(decoders_hash, GUINT_TO_POINTER(payload_type));
-	if (!decoder) {  /* Put either valid or empty decoder into the hash table */
-		decoder = g_new(rtp_decoder_t,1);
-		decoder->handle = NULL;
-		decoder->context = NULL;
-
-		if (rp->info->info_payload_type_str && find_codec(rp->info->info_payload_type_str)) {
-			p = rp->info->info_payload_type_str;
-		} else {
-			p = try_val_to_str_ext(payload_type, &rtp_payload_type_short_vals_ext);
-		}
-
-		if (p) {
-			decoder->handle = find_codec(p);
-			if (decoder->handle)
-				decoder->context = codec_init(decoder->handle);
-		}
-		g_hash_table_insert(decoders_hash, GUINT_TO_POINTER(payload_type), decoder);
-	}
-	if (decoder->handle) {  /* Decode with registered codec */
-		tmp_buff_len = codec_decode(decoder->handle, decoder->context, rp->payload_data, rp->info->info_payload_len, NULL, NULL);
-		tmp_buff = (SAMPLE *)g_malloc(tmp_buff_len);
-		decoded_bytes = codec_decode(decoder->handle, decoder->context, rp->payload_data, rp->info->info_payload_len, tmp_buff, &tmp_buff_len);
-		*out_buff = tmp_buff;
-
-		channels = codec_get_channels(decoder->handle, decoder->context);
-		sample_rate = codec_get_frequency(decoder->handle, decoder->context);
-
-		return decoded_bytes;
-	}
-
-	*out_buff = NULL;
-	return 0;
-}
-
-/****************************************************************************/
 static void
 update_progress_bar(gfloat fraction)
 {
@@ -633,7 +552,7 @@ decode_rtp_stream(rtp_stream_info_t *rsi, gpointer ptr)
 #endif
 	seq = 0;
 	start_timestamp = 0;
-	decoders_hash = g_hash_table_new_full(g_direct_hash, g_direct_equal, NULL, rtp_decoder_value_destroy);
+	decoders_hash = rtp_decoder_hash_table_new();
 
 	/* we update the progress bar 100 times */
 
@@ -671,7 +590,7 @@ decode_rtp_stream(rtp_stream_info_t *rsi, gpointer ptr)
 			seq = rp->info->info_seq_num - 1;
 		}
 
-		decoded_bytes = decode_rtp_packet(rp, &out_buff, decoders_hash);
+		decoded_bytes = decode_rtp_packet(rp, &out_buff, decoders_hash, &channels, &sample_rate);
 		if (decoded_bytes == 0) {
 			seq = rp->info->info_seq_num;
 		}
diff --git a/ui/rtp_media.c b/ui/rtp_media.c
new file mode 100644
index 0000000000..93f1f0b8ea
--- /dev/null
+++ b/ui/rtp_media.c
@@ -0,0 +1,134 @@
+/* rtp_media.c
+ *
+ * RTP decoding routines for Wireshark.
+ * Copied from ui/gtk/rtp_player.c
+ *
+ * Copyright 2006, Alejandro Vaquero <alejandrovaquero@yahoo.com>
+ *
+ * Wireshark - Network traffic analyzer
+ * By Gerald Combs <gerald@wireshark.org>
+ * 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+
+#include <codecs/codecs.h>
+
+#include <epan/rtp_pt.h>
+#include <epan/dissectors/packet-rtp.h>
+
+#include <ui/rtp_media.h>
+
+/****************************************************************************/
+/* DECODING */
+/****************************************************************************/
+
+typedef struct _rtp_decoder_t {
+    codec_handle_t handle;
+    void *context;
+} rtp_decoder_t;
+
+/****************************************************************************/
+/*
+ * Return the number of decoded bytes
+ */
+
+size_t
+decode_rtp_packet(rtp_packet_t *rp, SAMPLE **out_buff, GHashTable *decoders_hash, unsigned *channels_ptr, unsigned *sample_rate_ptr)
+{
+    unsigned int  payload_type;
+    const gchar *p;
+    rtp_decoder_t *decoder;
+    SAMPLE *tmp_buff = NULL;
+    size_t tmp_buff_len;
+    size_t decoded_bytes = 0;
+
+    if ((rp->payload_data == NULL) || (rp->info->info_payload_len == 0) ) {
+        return 0;
+    }
+
+    payload_type = rp->info->info_payload_type;
+
+    /* Look for registered codecs */
+    decoder = (rtp_decoder_t *)g_hash_table_lookup(decoders_hash, GUINT_TO_POINTER(payload_type));
+    if (!decoder) {  /* Put either valid or empty decoder into the hash table */
+        decoder = g_new(rtp_decoder_t,1);
+        decoder->handle = NULL;
+        decoder->context = NULL;
+
+        if (rp->info->info_payload_type_str && find_codec(rp->info->info_payload_type_str)) {
+            p = rp->info->info_payload_type_str;
+        } else {
+            p = try_val_to_str_ext(payload_type, &rtp_payload_type_short_vals_ext);
+        }
+
+        if (p) {
+            decoder->handle = find_codec(p);
+            if (decoder->handle)
+                decoder->context = codec_init(decoder->handle);
+        }
+        g_hash_table_insert(decoders_hash, GUINT_TO_POINTER(payload_type), decoder);
+    }
+    if (decoder->handle) {  /* Decode with registered codec */
+        tmp_buff_len = codec_decode(decoder->handle, decoder->context, rp->payload_data, rp->info->info_payload_len, NULL, NULL);
+        tmp_buff = (SAMPLE *)g_malloc(tmp_buff_len);
+        decoded_bytes = codec_decode(decoder->handle, decoder->context, rp->payload_data, rp->info->info_payload_len, tmp_buff, &tmp_buff_len);
+        *out_buff = tmp_buff;
+
+        if (channels_ptr) {
+            *channels_ptr = codec_get_channels(decoder->handle, decoder->context);
+        }
+
+        if (sample_rate_ptr) {
+            *sample_rate_ptr = codec_get_frequency(decoder->handle, decoder->context);
+        }
+
+        return decoded_bytes;
+    }
+
+    *out_buff = NULL;
+    return 0;
+}
+
+/****************************************************************************/
+static void
+rtp_decoder_value_destroy(gpointer dec_arg)
+{
+    rtp_decoder_t *dec = (rtp_decoder_t *)dec_arg;
+
+    if (dec->handle)
+        codec_release(dec->handle, dec->context);
+    g_free(dec_arg);
+}
+
+GHashTable *rtp_decoder_hash_table_new(void)
+{
+    return g_hash_table_new_full(g_direct_hash, g_direct_equal, NULL, rtp_decoder_value_destroy);
+}
+
+/*
+ * Editor modelines  -  http://www.wireshark.org/tools/modelines.html
+ *
+ * Local variables:
+ * c-basic-offset: 4
+ * tab-width: 8
+ * indent-tabs-mode: nil
+ * End:
+ *
+ * vi: set shiftwidth=4 tabstop=8 expandtab:
+ * :indentSize=4:tabSize=8:noTabs=true:
+ */
diff --git a/ui/rtp_media.h b/ui/rtp_media.h
new file mode 100644
index 0000000000..444c1db18c
--- /dev/null
+++ b/ui/rtp_media.h
@@ -0,0 +1,88 @@
+/* rtp_media.h
+ *
+ * RTP decoding routines for Wireshark.
+ * Copied from ui/gtk/rtp_player.c
+ *
+ * Copyright 2006, Alejandro Vaquero <alejandrovaquero@yahoo.com>
+ *
+ * Wireshark - Network traffic analyzer
+ * By Gerald Combs <gerald@wireshark.org>
+ * 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef __RTP_MEDIA_H__
+#define __RTP_MEDIA_H__
+
+/** @file
+ *  "RTP Player" dialog box common routines.
+ *  @ingroup main_ui_group
+ */
+
+#ifdef __cplusplus
+extern "C" {
+#endif /* __cplusplus */
+
+#include <glib.h>
+
+/****************************************************************************/
+/* INTERFACE */
+/****************************************************************************/
+
+typedef gint16 SAMPLE;
+
+/* Defines an RTP packet */
+typedef struct _rtp_packet {
+    struct _rtp_info *info;	/* the RTP dissected info */
+    double arrive_offset;	/* arrive offset time since the beginning of the stream in ms */
+    guint8* payload_data;
+} rtp_packet_t;
+
+/** Create a new hash table.
+ *
+ * @return A new hash table suitable for passing to decode_rtp_packet.
+ */
+GHashTable *rtp_decoder_hash_table_new(void);
+
+/** Decode an RTP packet
+ *
+ * @param rp Wrapper for per-packet RTP tap data.
+ * @param out_buff Output audio samples.
+ * @param decoders_hash Hash table created with rtp_decoder_hash_table_new.
+ * @param channels_ptr If non-NULL, receives the number of channels in the sample.
+ * @param sample_rate_ptr If non-NULL, receives the sample rate.
+ * @return The number of decoded bytes on success, 0 on failure.
+ */
+size_t decode_rtp_packet(rtp_packet_t *rp, SAMPLE **out_buff, GHashTable *decoders_hash, unsigned *channels_ptr, unsigned *sample_rate_ptr);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+#endif /* __RTP_MEDIA_H__ */
+
+/*
+ * Editor modelines  -  http://www.wireshark.org/tools/modelines.html
+ *
+ * Local variables:
+ * c-basic-offset: 4
+ * tab-width: 8
+ * indent-tabs-mode: nil
+ * End:
+ *
+ * vi: set shiftwidth=4 tabstop=8 expandtab:
+ * :indentSize=4:tabSize=8:noTabs=true:
+ */