From e4ba6dbc3f1e76890b22773807ea37fe8fa2b1bc Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Wed, 10 Apr 2024 22:34:10 +0200 Subject: Adding upstream version 4.2.2. Signed-off-by: Daniel Baumann --- ui/rtp_media.h | 90 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 90 insertions(+) create mode 100644 ui/rtp_media.h (limited to 'ui/rtp_media.h') diff --git a/ui/rtp_media.h b/ui/rtp_media.h new file mode 100644 index 00000000..2de2a853 --- /dev/null +++ b/ui/rtp_media.h @@ -0,0 +1,90 @@ +/** @file + * + * RTP decoding routines for Wireshark. + * Copied from ui/gtk/rtp_player.c + * + * Copyright 2006, Alejandro Vaquero + * + * Wireshark - Network traffic analyzer + * By Gerald Combs + * Copyright 1999 Gerald Combs + * + * SPDX-License-Identifier: GPL-2.0-or-later + */ + +#ifndef __RTP_MEDIA_H__ +#define __RTP_MEDIA_H__ + +#include +#include + +/** @file + * "RTP Player" dialog box common routines. + * @ingroup main_ui_group + */ + +#ifdef __cplusplus +extern "C" { +#endif /* __cplusplus */ + +/****************************************************************************/ +/* INTERFACE */ +/****************************************************************************/ + +typedef gint16 SAMPLE; +#define SAMPLE_MAX G_MAXINT16 +#define SAMPLE_MIN G_MININT16 +#define SAMPLE_NaN SAMPLE_MIN +#define SAMPLE_BYTES (sizeof(SAMPLE) / sizeof(char)) + +/* Defines an RTP packet */ +typedef struct _rtp_packet { + guint32 frame_num; /* Qt only */ + struct _rtp_info *info; /* the RTP dissected info */ + double arrive_offset; /* arrive offset time since the beginning of the stream as ms in GTK UI and s in Qt UI */ + 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 payload from an RTP packet + * For RTP packets with dynamic payload types, the payload name, clock rate, + * and number of audio channels (e.g., from the SDP) can be provided. + * Note that the output sample rate and number of channels might not be the + * same as that of the input. + * + * @param payload_type Payload number + * @param payload_type_str Payload name, can be NULL + * @param payload_rate Sample rate, can be 0 for codec default + * @param payload_channels Audio channels, can be 0 for codec default + * @param payload_fmtp_map Map of format parameters for the media type + * @param payload_data Payload + * @param payload_len Length of payload + * @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_payload(guint8 payload_type, const gchar *payload_type_str, int payload_rate, int payload_channels, wmem_map_t *payload_fmtp_map, guint8 *payload_data, size_t payload_len, SAMPLE **out_buff, GHashTable *decoders_hash, guint *channels_ptr, guint *sample_rate_ptr); + +/** 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, guint *channels_ptr, guint *sample_rate_ptr); + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + +#endif /* __RTP_MEDIA_H__ */ -- cgit v1.2.3