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wireshark
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Correct language. 

svn path=/trunk/; revision=27060
This commit is contained in:
Jaap Keuter 2008-12-19 11:31:28 +00:00
parent 0c7d516955
commit c95944a16a
1 changed files with 13 additions and 13 deletions
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26
gtk/rtp_player.c
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@ -29,23 +29,23 @@
* packet
* - add_rtp_packet() will add the RTP packet in a RTP stream struct, and
* create the RTP stream if it is the first RTP in the stream.
* - Each new RTP stream will be added to a list of RTP stream, called
* - Each new RTP stream will be added to a list of RTP streams, called
* rtp_streams_list
* - When the user clicks "Player" in the VoipCall dialogue,
* rtp_player_init() is called.
* - rtp_player_init() create the main dialog, and it calls:
* + mark_rtp_stream_to_play() to mark the RTP streams that needs to be
* displayed. These are the RTP stream that match the selected calls in
* displayed. These are the RTP streams that match the selected calls in
* the VoipCall dlg.
* + decode_rtp_stream() this will decode the RTP packets in each RTP
* stream, and will also create the RTP channles. An RTP channel is a
* group of RTP stream that have in common the source and destination
* IP and UPD ports. The RTP channels is what the user will listen in
* one of the two Audio channles.
* stream, and will also create the RTP channels. An RTP channel is a
* group of RTP streams that have in common the source and destination
* IP and UDP ports. The RTP channels is what the user will listen in
* one of the two Audio channels.
* The RTP channels are stored in the hash table rtp_channels_hash
* + add_channel_to_window() will create and add the Audio graphic
* representation in the main window
* - When the user click the check box to listen one of the Audio channels,
* - When the user clicks the check box to listen one of the Audio channels,
* the structure rtp_channels is filled to play one or two RTP channels
* (a max of two channels can be listened at a given moment)
*/
@ -232,10 +232,10 @@ typedef struct _rtp_play_channles {
PaTime out_diff_time;
PaTime pa_start_time;
#endif /* PORTAUDIO_API_1 */
} rtp_play_channles_t;
} rtp_play_channels_t;
/* The two RTP channles to play */
static rtp_play_channles_t *rtp_channels = NULL;
static rtp_play_channels_t *rtp_channels = NULL;
typedef struct _rtp_decoder_t {
codec_handle_t handle;
@ -388,7 +388,7 @@ add_rtp_packet(const struct _rtp_info *rtp_info, packet_info *pinfo)
g_hash_table_insert(rtp_streams_hash, g_strdup(key_str->str), stream_info);
/* Add the element to the List too. The List is used to decode the packets because it is sordted */
/* Add the element to the List too. The List is used to decode the packets because it is sorted */
rtp_streams_list = g_list_append(rtp_streams_list, stream_info);
}
@ -1038,7 +1038,7 @@ draw_cursors(gpointer data _U_)
static void
init_rtp_channels_vals(void)
{
rtp_play_channles_t *rpci = rtp_channels;
rtp_play_channels_t *rpci = rtp_channels;
/* if we only have one channel to play, we just use the info from that channel */
if (rpci->rci[0] == NULL) {
@ -1097,7 +1097,7 @@ static int paCallback( const void *inputBuffer, void *outputBuffer,
void *userData)
{
#endif /* PORTAUDIO_API_1 */
rtp_play_channles_t *rpci = (rtp_play_channles_t*)userData;
rtp_play_channels_t *rpci = (rtp_play_channels_t *)userData;
SAMPLE *wptr = (SAMPLE*)outputBuffer;
sample_t sample;
unsigned int i;
@ -2016,7 +2016,7 @@ rtp_player_init(voip_calls_tapinfo_t *voip_calls_tap)
#endif /* HAVE_G729_G723 */
if (!rtp_channels) {
rtp_channels = g_malloc(sizeof(rtp_play_channles_t));
rtp_channels = g_malloc(sizeof(rtp_play_channels_t));
}
reset_rtp_channels();