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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>
This commit is contained in:
Gerald Combs 2015-09-18 15:13:29 -07:00
parent 723bae5eff
commit 587e93a55e
5 changed files with 234 additions and 89 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
ui/CMakeLists.txt
View File

@ -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

2
ui/Makefile.common
View File

@ -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 \

97
ui/gtk/rtp_player.c
View File

@ -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;
@ -274,17 +260,6 @@ rtp_stream_value_destroy(gpointer rsi_arg)
rsi = NULL;
}
/****************************************************************************/
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 )
@ -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);
@ -453,62 +428,6 @@ mark_all_rtp_stream_to_decode(gchar *key _U_ , rtp_stream_info_t *rsi, gpointer
total_packets += rsi->packet_count;
}
/****************************************************************************/
/* 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;
}

134
ui/rtp_media.c Normal file
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@ -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:
*/

88
ui/rtp_media.h Normal file
View File

@ -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:
*/