Age | Commit message (Collapse) | Author | Files | Lines |
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svn path=/trunk/; revision=31731
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gtk_progress_bar_update;
svn path=/trunk/; revision=31384
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default Host API (go figure).
Therefore opening the default stream may fail in the precense of usable devices, on other Host API's.
If the default stream fails to open iterate among the Host API's to find one with a default device.
svn path=/trunk/; revision=31318
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svn path=/trunk/; revision=31298
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svn path=/trunk/; revision=31297
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Add RTP player to "RTP Stream Analysis" and makeover of VoIP flow
https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=4240
svn path=/trunk/; revision=30967
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svn path=/trunk/; revision=29468
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level window.
svn path=/trunk/; revision=28810
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When audio samples have to be dropped or silence samples inserted to reflect
the timestamp there is no indication of these problems on the display.
I propose that such problems be indicated on the waveform display by the use of
amber coloration and that the number of incorrect timestamps be listed
svn path=/trunk/; revision=28451
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svn path=/trunk/; revision=28358
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library.
At least that is what the documentation says.
svn path=/trunk/; revision=27073
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svn path=/trunk/; revision=27060
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svn path=/trunk/; revision=26873
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equivalent API
gtk_button_box_set_spacing() ==> gtk_box_set_spacing()
svn path=/trunk/; revision=25673
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equivalent API
gtk_container_border_width() ==> gtk_container_set_border_width()
gtk_container_children() ==> gtk_container_get_children()
gtk_entry_new_with_max_length() ==> gtk_entry_new(); gtk_entry_set_max_length()
gtk_menu_append() ==> gtk_menu_shell_append()
gtk_menu_prepend() ==> gtk_menu_shell_prepend()
gtk_notebook_set_page() ==> gtk_notebook_set_current_page()
gtk_paned_gutter_size() ==> gtk_paned_set_gutter_size()
gtk_radio_button_group() ==> gtk_radio_button_get_group()
gtk_signal_connect() ==> g_signal_connect()
gtk_signal_disconnect() ==> g_signal_handler_disconnect()
gtk_signal_emit_by_name() ==> g_signal_emit_by_name()
gtk_signal_handler_block_by_data() ==> g_signal_handlers_block_matched()
gtk_signal_handler_block_by_func() ==> g_signal_handlers_block_by_func()
gtk_signal_handler_unblock_by_data() ==> g_signal-handlers_unblock_matched()
gtk_signal_handler_unblock_by_func() ==> g_signal-handlers_unblock_by_func()
gtk_spin_button_get_value_as_float() ==> gtk_spin_button_get_value()
gtk_toggle_button_set_state() ==> gtk_toggle_button_set_active()
svn path=/trunk/; revision=25634
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Added a new checkbox for the RTP player to use the RTP timing instead of the
arriving time of the packets. This is useful when the RTP is being tunneled
(e.g. EtherIP), so the original timing is missing.
In some cases when encapsulating the original IP/RTP over EtherIP, the
encapsulated data comes in chunks of 1sec, so it is useful to use the RTP
timing to play it. In these cases we can not simulate jitter buffer, so this is
disabled when checking the new "Use RTP timestamp" option.
svn path=/trunk/; revision=25293
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svn path=/trunk/; revision=24969
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svn path=/trunk/; revision=24922
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svn path=/trunk/; revision=24918
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svn path=/trunk/; revision=24911
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svn path=/trunk/; revision=24910
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gtk_button_new_from_stock
svn path=/trunk/; revision=24904
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svn path=/trunk/; revision=24896
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svn path=/trunk/; revision=24823
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This patch provides a new RTP Player preferences dialog. It allows one to
select the maximum number of visable channels in the RTP Player window. The
default is four (4) channels which is the maximum number of usable channels
that the RTP Player can display and still have access to the bottom row buttons
on a 1024*768 resolution display. Specifying a value less than 1 or greater than
10 will be result in the RTP Player displaying the default 4 channels.
svn path=/trunk/; revision=24112
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two different variables for types voip_rtp_stream_info_t (strinfo) and
voip_calls_info_t (callsinfo) instead of using strinfo for both.
svn path=/trunk/; revision=24038
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svn path=/trunk/; revision=24032
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svn path=/trunk/; revision=23322
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new codec plugin type
search registered codecs in rtp player
fix memory leak in rtp player
svn path=/trunk/; revision=23270
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svn path=/trunk/; revision=23240
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Voip graph and RTP player windows. Make the calls selection using standard multiple selection using the GTK buildin functions. Add a 'Select All' buttom to the VoIP calls window.
svn path=/trunk/; revision=23089
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svn path=/trunk/; revision=22549
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Note that we should return an error for payload types we don't support,
so the user knows we can't handle the audio.
svn path=/trunk/; revision=22547
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svn path=/trunk/; revision=22485
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svn path=/trunk/; revision=22479
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svn path=/trunk/; revision=21444
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svn path=/trunk/; revision=21439
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svn path=/trunk/; revision=21436
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svn path=/trunk/; revision=21080
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null out the pointer to it. This prevents a freed pixbuf from being
used.
svn path=/trunk/; revision=19687
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Fix typos.
Null out "pa_stream" when the stream is closed; this appears to prevent
some crashes.
svn path=/trunk/; revision=19633
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svn path=/trunk/; revision=19632
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it supports NULL for "no text".
svn path=/trunk/; revision=19631
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created yet.
svn path=/trunk/; revision=19630
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svn path=/trunk/; revision=19629
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Here is a patch for spelling typos in comments and strings in the gtk/
directories.
svn path=/trunk/; revision=19568
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Add portaudio v19 support in addition to v18:
Here is a patch using the API 2 (v19). I tested it only in Windows.
Attached also the Windows v19 zip file used. I did not modify the
Makefile.nmake to download this new version (when running the "setup"),
because Windows v18 it is still good.
svn path=/trunk/; revision=19446
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player, so that we get declartions of rtp_player_init() and
add_rtp_packet().
Constify the first argument to add_rtp_packet(), as it's passed a
pointer to a const value.
svn path=/trunk/; revision=19272
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actually do something useful except to print that the new API is not
supported.
Adapt acinclude.m4 to provide PORTAUDIO_API_1 and add back the
check_message to indicate *why* we are checking for the type.
svn path=/trunk/; revision=19148
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svn path=/trunk/; revision=19144
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