forked from osmocom/wireshark
8186ab3d9f
Small bugs were introduced when copy/pasting the code from GTK UI: - arrive_offset is stored in seconds and not milliseconds - some tests regarding the current playback mode were wrong Change-Id: I21fb82ba8ff6c8defa7df90c815c040e9e074aaa Reviewed-on: https://code.wireshark.org/review/13885 Petri-Dish: Pascal Quantin <pascal.quantin@gmail.com> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Pascal Quantin <pascal.quantin@gmail.com>
92 lines
2.8 KiB
C
92 lines
2.8 KiB
C
/* rtp_media.h
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*
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* RTP decoding routines for Wireshark.
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* Copied from ui/gtk/rtp_player.c
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*
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* Copyright 2006, Alejandro Vaquero <alejandrovaquero@yahoo.com>
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*
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* Wireshark - Network traffic analyzer
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* By Gerald Combs <gerald@wireshark.org>
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* Copyright 1999 Gerald Combs
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#ifndef __RTP_MEDIA_H__
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#define __RTP_MEDIA_H__
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/** @file
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* "RTP Player" dialog box common routines.
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* @ingroup main_ui_group
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*/
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#ifdef __cplusplus
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extern "C" {
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#endif /* __cplusplus */
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#include <glib.h>
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/****************************************************************************/
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/* INTERFACE */
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/****************************************************************************/
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typedef gint16 SAMPLE;
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#define SAMPLE_MAX G_MAXINT16
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#define SAMPLE_MIN G_MININT16
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/* Defines an RTP packet */
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typedef struct _rtp_packet {
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guint32 frame_num; /* Qt only */
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struct _rtp_info *info; /* the RTP dissected info */
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double arrive_offset; /* arrive offset time since the beginning of the stream as ms in GTK UI and s in Qt UI */
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guint8* payload_data;
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} rtp_packet_t;
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/** Create a new hash table.
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*
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* @return A new hash table suitable for passing to decode_rtp_packet.
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*/
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GHashTable *rtp_decoder_hash_table_new(void);
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/** Decode an RTP packet
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*
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* @param rp Wrapper for per-packet RTP tap data.
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* @param out_buff Output audio samples.
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* @param decoders_hash Hash table created with rtp_decoder_hash_table_new.
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* @param channels_ptr If non-NULL, receives the number of channels in the sample.
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* @param sample_rate_ptr If non-NULL, receives the sample rate.
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* @return The number of decoded bytes on success, 0 on failure.
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*/
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size_t decode_rtp_packet(rtp_packet_t *rp, SAMPLE **out_buff, GHashTable *decoders_hash, unsigned *channels_ptr, unsigned *sample_rate_ptr);
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#ifdef __cplusplus
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}
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#endif /* __cplusplus */
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#endif /* __RTP_MEDIA_H__ */
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/*
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* Editor modelines - http://www.wireshark.org/tools/modelines.html
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*
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* Local variables:
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* c-basic-offset: 4
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* tab-width: 8
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* indent-tabs-mode: nil
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* End:
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*
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* vi: set shiftwidth=4 tabstop=8 expandtab:
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* :indentSize=4:tabSize=8:noTabs=true:
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*/
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