Adding upstream version 4.2.2.upstream/4.2.2

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 90 insertions, 0 deletions

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 <alejandrovaquero@yahoo.com>
+ *
+ * Wireshark - Network traffic analyzer
+ * By Gerald Combs <gerald@wireshark.org>
+ * Copyright 1999 Gerald Combs
+ *
+ * SPDX-License-Identifier: GPL-2.0-or-later
+ */
+
+#ifndef __RTP_MEDIA_H__
+#define __RTP_MEDIA_H__
+
+#include <glib.h>
+#include <wsutil/wmem/wmem_map.h>
+
+/** @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__ */