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wireshark/ui/qt/rtp_player_dialog.h

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Add UI header files to Doxygen Add @file markers for UI header files so that Doxygen will generate documentation for them.
2021-11-30 13:01:36 +00:00
/** @file
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
*
* Wireshark - Network traffic analyzer
* By Gerald Combs <gerald@wireshark.org>
* Copyright 1998 Gerald Combs
*
Fix comment end after SPDX identifier Move */ to a separate line below the SPDX identifier. Change-Id: Id1032215449cfccae0933147b45e04b65e0b727f Reviewed-on: https://code.wireshark.org/review/27211 Reviewed-by: Anders Broman <a.broman58@gmail.com>
2018-04-30 07:47:58 +00:00
* SPDX-License-Identifier: GPL-2.0-or-later
*/
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
#ifndef RTP_PLAYER_DIALOG_H
#define RTP_PLAYER_DIALOG_H
#include "config.h"
#include <glib.h>
VoIP Dialogs: Refactoring of singleton windows to factory methods Singletons moved from main_window to each class's static open<NameOfClass> method: - RtpPlayerDialog - RtpStreamDialog - VoipCallsDialog - RtpAnalysisDialog Fixed issue with selecting RTP stream in sequence dialog. When user selected a stream and moved mouse to Rtp Player button and pressed it, incorrect RTP stream was sent to it.
2021-05-09 14:29:11 +00:00
#include <mutex>
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
#include "ui/rtp_stream.h"
#include "wireshark_dialog.h"
RTP Player dialog improved Functional changes: Audio routing information is now stored in audio stream and not in table. It is handled by separate utils/rtp_audio_routing.cpp class which is able to convert it between mono/stereo etc. There is new utils/rtp_audio_routing_filter.cpp class which is able to route mono audio stream to any audio channel. Sample file separated from audio stream file - sample file is generated only during recap. So when we need new waveform, we just use existing sample file and no recap required - it is much faster. Audio stream exports just mono audio. Mono audio is then expanded to stereo and correct channel by AudioRoutingFilter during play as required. So when audio routing is changed, no recap and no audio export is required. When audio stream is muted, no audio is produced nor played. Most of signals between RtpPlayerDialog and RtpAudioStream were removed. Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams). Play possition is not received from every playing stream but from independent silence stream. Added Mute/Unmute function. Optimalization: When audio routing is changed, just graphs are updated. No retap nor audio decoding is required. When TOD is changed, just graphs are updated. No retap nor audio decoding is required.
2021-02-08 21:35:12 +00:00
#include "rtp_audio_stream.h"
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
VoIP Dialogs: Refactoring of singleton windows to factory methods Singletons moved from main_window to each class's static open<NameOfClass> method: - RtpPlayerDialog - RtpStreamDialog - VoipCallsDialog - RtpAnalysisDialog Fixed issue with selecting RTP stream in sequence dialog. When user selected a stream and moved mouse to Rtp Player button and pressed it, incorrect RTP stream was sent to it.
2021-05-09 14:29:11 +00:00
#include <QWidget>
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
#include <QMap>
VoIP dialogs: Performance improvements Retap and UI response are much faster when many RTP streams are processed. RTP Streams/Analyse 1000+, RTP Player 500+. Changes: - RTP streams are searched with hash, not by iterating over list. - UI operations do not redraw screen after every change, just after all changes. UI is locked when rereading packets. - Sample list during RTP decoding is stored in memory so wireshark uses just half of opened files for audio decoding than before. - Analysis window checkbox area is limited in height - Dialogs shows shows count of streams, count of selected streams and count of unmuted streams - Documentation extended with chapter about RTP decoding parameters - Documentation extended with performance estimates
2021-04-13 14:38:13 +00:00
#include <QMultiHash>
rtp_player_dialog: Route audio for a stream to left/right speaker in RTP player Column 'Play' added to player. Double click on a stream in the column changes audio routing for the stream. When soundcard supports only one channel, there are Mute/Play option. When soundcard supports two or more channels, there are Mute/L/L+R/R options. Muted channel is drawn with dotted line. Change-Id: If120c902195da46f98a1663c589f20c6a1da0ba7 Reviewed-on: https://code.wireshark.org/review/35687 Petri-Dish: Alexis La Goutte <alexis.lagoutte@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-07 21:16:42 +00:00
#include <QTreeWidgetItem>
RTP Player: Audio routing information refactored Audio routing information storage refactored to separate mute information and channel routing. It is required for next changes.
2021-02-04 13:12:44 +00:00
#include <QMetaType>
RTP Player: Selected waveform is emphased. Waveform can be selected by click. When there is many waveforms, it is not clear which one is selected in list. It is important for selecting packets from graph. Patch emphasis selected waveform. In addition, it allows to select waveform by clicking on it.
2021-02-04 13:23:43 +00:00
#include <ui/qt/widgets/qcustomplot.h>
Readd feature to make QtXMultimedia optional again In order to be able to defer solving all Qt6 API differences at once I tried to reactivate the QT_MULTIMEDIA_LIB feature. I managed to fix most problems but one problem remains in both Qt5 and Qt6 builds. Without Qt[56]Multimedia, the following error exceeds my non-existing C++ knowledge: jmayer/work/wireshark/git/ui/qt/rtp_player_dialog.cpp:154:18: error: out-of-line definition of 'RtpPlayerDialog' does not match any declaration in 'RtpPlayerDialog' RtpPlayerDialog::RtpPlayerDialog(QWidget &parent, CaptureFile &cf, bool capture_running) : ^~~~~~~~~~~~~~~
2021-11-21 21:05:36 +00:00
#ifdef QT_MULTIMEDIA_LIB
Qt: Update the RTP stream UI to support Qt6Multimedia. Fixes #18115
2022-05-24 01:59:35 +00:00
# if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
# include <QAudioDevice>
# else
# include <QAudioDeviceInfo>
# endif
Readd feature to make QtXMultimedia optional again In order to be able to defer solving all Qt6 API differences at once I tried to reactivate the QT_MULTIMEDIA_LIB feature. I managed to fix most problems but one problem remains in both Qt5 and Qt6 builds. Without Qt[56]Multimedia, the following error exceeds my non-existing C++ knowledge: jmayer/work/wireshark/git/ui/qt/rtp_player_dialog.cpp:154:18: error: out-of-line definition of 'RtpPlayerDialog' does not match any declaration in 'RtpPlayerDialog' RtpPlayerDialog::RtpPlayerDialog(QWidget &parent, CaptureFile &cf, bool capture_running) : ^~~~~~~~~~~~~~~
2021-11-21 21:05:36 +00:00
#endif
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
namespace Ui {
class RtpPlayerDialog;
}
class QCPItemStraightLine;
Qt: Add a play button to the RTP Stream Analysis dialog. Rename the "Play Call" button to "Play Streams". Move the button creation code to a common routine. Use it to add a "Play Streams" button to the RTP Stream Analysis, similar to the GTK+ UI. Don't restrict RTP to IPv[46] as suggested by Michal. I don't have any RTP-over-Bluetooth captures so I can't test this directly. Change-Id: I4703cac1d5bf5b3ff0255d36da2c5164feb0547d Reviewed-on: https://code.wireshark.org/review/10888 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-10-08 18:31:42 +00:00
class QDialogButtonBox;
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
class QMenu;
class RtpAudioStream;
Update to qcustomplot 2.0.1 Upgrade the internal version of QCustomPlot to 2.0.1 Change-Id: I1eb372d8e6a2f6c1bbdde4c74596785bf2d405c2 Reviewed-on: https://code.wireshark.org/review/17980 Reviewed-by: Roland Knall <rknall@gmail.com> Petri-Dish: Roland Knall <rknall@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2016-09-25 09:57:55 +00:00
class QCPAxisTicker;
class QCPAxisTickerDateTime;
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
RTP Player: Added ability to save streams as .au and .wav Features: - saves multiple streams (all selected and unmuted) - saves streams same way they are played (jitter buffer, sampling, ...) - only streams with audio (play rate >0) are exported - streams with play rate == 0 are silently ignored even selected for export - all exported streams must use same play rate (user can change it before save)
2021-03-31 14:52:06 +00:00
typedef enum {
save_audio_none,
save_audio_au,
save_audio_wav
} save_audio_t;
RTP Player: Added ability to save streams as .au and .wav Features: - saves multiple streams (all selected and unmuted) - saves streams same way they are played (jitter buffer, sampling, ...) - only streams with audio (play rate >0) are exported - streams with play rate == 0 are silently ignored even selected for export - all exported streams must use same play rate (user can change it before save)
2021-04-01 08:07:22 +00:00
typedef enum {
save_payload_none,
save_payload_data
} save_payload_t;
RTP Player: Export of audio can start at position of play cursor Added additional option for audio exporting 'From cursor'. Documentation updated.
2021-04-27 20:01:16 +00:00
typedef enum {
save_mode_from_cursor,
save_mode_sync_stream,
save_mode_sync_file
} save_mode_t;
VoIP Dialogs: Refactoring of singleton windows to factory methods Singletons moved from main_window to each class's static open<NameOfClass> method: - RtpPlayerDialog - RtpStreamDialog - VoipCallsDialog - RtpAnalysisDialog Fixed issue with selecting RTP stream in sequence dialog. When user selected a stream and moved mouse to Rtp Player button and pressed it, incorrect RTP stream was sent to it.
2021-05-09 14:29:11 +00:00
// Singleton by https://refactoring.guru/design-patterns/singleton/cpp/example#example-1
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
class RtpPlayerDialog : public WiresharkDialog
{
Q_OBJECT
Qt: Fix currentOutputDeviceName() without QtMultimedia Fix build error: ui/qt/moc_rtp_player_dialog.cxx:87:76: error: ‘currentOutputDeviceName’ was not declared in this scope case 0: *reinterpret_cast< QString*>(_v) = currentOutputDeviceName(); break; Change-Id: I065862540e775c3e965cb5d3ae4c53bd8d505bdd Reviewed-on: https://code.wireshark.org/review/19142 Petri-Dish: Michal Labedzki <michal.labedzki@tieto.com> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Michael Mann <mmann78@netscape.net>
2016-12-08 07:14:14 +00:00
#ifdef QT_MULTIMEDIA_LIB
Qt: Fixup the currentOutputDeviceName Q_PROPERTY. The CONSTANT attribute indicates that the same value will be returned every time. That isn't the case here so remove it. Change-Id: Ie7451e6aabcb4fa1a6960762d96ad190f32b3d7a Reviewed-on: https://code.wireshark.org/review/19130 Reviewed-by: Gerald Combs <gerald@wireshark.org> Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Michael Mann <mmann78@netscape.net>
2016-12-07 17:58:28 +00:00
Q_PROPERTY(QString currentOutputDeviceName READ currentOutputDeviceName)
Qt: Fix currentOutputDeviceName() without QtMultimedia Fix build error: ui/qt/moc_rtp_player_dialog.cxx:87:76: error: ‘currentOutputDeviceName’ was not declared in this scope case 0: *reinterpret_cast< QString*>(_v) = currentOutputDeviceName(); break; Change-Id: I065862540e775c3e965cb5d3ae4c53bd8d505bdd Reviewed-on: https://code.wireshark.org/review/19142 Petri-Dish: Michal Labedzki <michal.labedzki@tieto.com> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Michael Mann <mmann78@netscape.net>
2016-12-08 07:14:14 +00:00
#endif
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
public:
VoIP Dialogs: Refactoring of singleton windows to factory methods Singletons moved from main_window to each class's static open<NameOfClass> method: - RtpPlayerDialog - RtpStreamDialog - VoipCallsDialog - RtpAnalysisDialog Fixed issue with selecting RTP stream in sequence dialog. When user selected a stream and moved mouse to Rtp Player button and pressed it, incorrect RTP stream was sent to it.
2021-05-09 14:29:11 +00:00
/**
* Returns singleton
*/
static RtpPlayerDialog *openRtpPlayerDialog(QWidget &parent, CaptureFile &cf, QObject *packet_list, bool capture_running);
/**
* Should not be clonnable and assignable
*/
RtpPlayerDialog(RtpPlayerDialog &other) = delete;
void operator=(const RtpPlayerDialog &) = delete;
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
Make sure we can compile without QtMultimedia. Change-Id: I8db453a735956435fc6e2e4276961adb1f7ed11a Reviewed-on: https://code.wireshark.org/review/10892 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-10-08 21:56:06 +00:00
/**
* @brief Common routine to add a "Play call" button to a QDialogButtonBox.
* @param button_box Caller's QDialogButtonBox.
* @return The new "Play call" button.
*/
VoIP dialogs: Default actions for buttons works as before When button is pressed or triggered by shortcut, it opens same window as before. User can click small arrow next to button and it open menu with all new actions e.g. Set/Add/Remove for RTP Player. Documentation updated.
2021-05-01 22:33:54 +00:00
static QToolButton *addPlayerButton(QDialogButtonBox *button_box, QDialog *dialog);
Make sure we can compile without QtMultimedia. Change-Id: I8db453a735956435fc6e2e4276961adb1f7ed11a Reviewed-on: https://code.wireshark.org/review/10892 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-10-08 21:56:06 +00:00
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
#ifdef QT_MULTIMEDIA_LIB
Fix crash in RTP Player on stop and close When dragging the UI, this somehow causes a great lag. Then by spam-clicking on the Stop button, a double free seems to occur. Fix this by moving the audio cleanup to the outputStateChanged callback as documented at https://doc.qt.io/qt-5/qaudiooutput.html. Note that calling stop() in the IdleState also triggers a change event, resulting in the desired cleanup. Stop streams before the dialog is closed (via accept/reject). This *cannot* be done in the destrutor of RtpPlayerDialog because destructing QAudioOutput processes events from the event queue, resulting in preature destruction of other objects... crash. Change-Id: I6bfb33c9396e9bc1ffd346519d22390a97b6bdaf Reviewed-on: https://code.wireshark.org/review/11894 Petri-Dish: Peter Wu <peter@lekensteyn.nl> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-11-17 17:54:41 +00:00
void accept();
void reject();
rtp_player: Player is able to set start of audio play by double click Patch adds ability to set start of audio play by double clicking on waveform. Patch fixes unreported issue with placing waveform at incorrect place when switched relative/absolute time mode (check/uncheck Time of Day). Change-Id: Ib8ce24aea870e2443e033afbb6d6e9fbcf222431 Reviewed-on: https://code.wireshark.org/review/35621 Petri-Dish: Anders Broman <a.broman58@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-01 23:38:35 +00:00
void setMarkers();
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
RTP Player: Dialog is nonmodal now and can be called multiple ways Changes: - refactored main_dialog handling of telephony dialogs - RTP Player dialog is nonmodal now and can be left open - it is possible to issue three actions on RTP Player dialog from other dialogs (other dialog have selected set of RTP streams before action) - replace - removes existing streams from RTP dialog and shows new set - add - adds new set to existing list in RTP dialog - remove - remove streams in set from list in RTP dialog - Sequence Dialog: - was modified to hold rtpstream_info_t for RTP streams - added Play button - VoIP features (RTP Play button, select/deselect RTP stream) are disabled after creation and must be enabled. It handles that RTP Play button is not shown e.g. in TCP sequence show
2021-03-27 20:01:43 +00:00
/** Replace/Add/Remove an RTP streams to play.
* Requires array of rtpstream_info_t.
* Each item must have filled items: src_addr, src_port, dest_addr,
* dest_port, ssrc, packet_count, setup_frame_number, and start_rel_time.
*
Qt: Fix some documentation warnings.
2021-04-30 22:23:35 +00:00
* @param stream_ids struct with rtpstream info
RTP Player: Dialog is nonmodal now and can be called multiple ways Changes: - refactored main_dialog handling of telephony dialogs - RTP Player dialog is nonmodal now and can be left open - it is possible to issue three actions on RTP Player dialog from other dialogs (other dialog have selected set of RTP streams before action) - replace - removes existing streams from RTP dialog and shows new set - add - adds new set to existing list in RTP dialog - remove - remove streams in set from list in RTP dialog - Sequence Dialog: - was modified to hold rtpstream_info_t for RTP streams - added Play button - VoIP features (RTP Play button, select/deselect RTP stream) are disabled after creation and must be enabled. It handles that RTP Play button is not shown e.g. in TCP sequence show
2021-03-27 20:01:43 +00:00
*/
RTP Player: Added button 'Refresh streams' for live capture
2021-04-22 19:33:11 +00:00
void replaceRtpStreams(QVector<rtpstream_id_t *> stream_ids);
void addRtpStreams(QVector<rtpstream_id_t *> stream_ids);
void removeRtpStreams(QVector<rtpstream_id_t *> stream_ids);
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
signals:
VoIP dialogs: Improvements and new functions Changes: - RTP Player added to Telephony/RTP menu. - When openning RTP Analysis or RTP Player from RTP menu, just selected stream is added. When Ctrl is hold during opening, reverse stream is searched and added too. - RTP Player: Added tool to select/deselect all inaudible streams - RTP Player: Added Prepare Filter button - RTP Player: Added Analyze button - RTP Analysis: Added Prepare Filter button - documentation updated Code changes: - RTP Player::rescanPacket() is not fired multiple times during rate change and during dialog creation - Error shown in RTP player is cleared after every new decode of streams - RTP Player handles case when Qt do not emit stop stream event - "Select" menu code unified between dialogs> - RTP Player: Audio routing menu unified - buttons are connected to actions by signals() - Analyze dialog is called by list of rtpstream_id, not rtpstream_info
2021-04-14 13:47:07 +00:00
// Tells the packet list to redraw. An alternative might be to add a
// cf_packet_marked callback to file.[ch] but that's synchronous and
// might incur too much overhead.
void packetsMarked();
void updateFilter(QString filter, bool force = false);
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
void goToPacket(int packet_num);
VoIP dialogs: Improvements and new functions Changes: - RTP Player added to Telephony/RTP menu. - When openning RTP Analysis or RTP Player from RTP menu, just selected stream is added. When Ctrl is hold during opening, reverse stream is searched and added too. - RTP Player: Added tool to select/deselect all inaudible streams - RTP Player: Added Prepare Filter button - RTP Player: Added Analyze button - RTP Analysis: Added Prepare Filter button - documentation updated Code changes: - RTP Player::rescanPacket() is not fired multiple times during rate change and during dialog creation - Error shown in RTP player is cleared after every new decode of streams - RTP Player handles case when Qt do not emit stop stream event - "Select" menu code unified between dialogs> - RTP Player: Audio routing menu unified - buttons are connected to actions by signals() - Analyze dialog is called by list of rtpstream_id, not rtpstream_info
2021-04-14 13:47:07 +00:00
void rtpAnalysisDialogReplaceRtpStreams(QVector<rtpstream_id_t *> stream_infos);
void rtpAnalysisDialogAddRtpStreams(QVector<rtpstream_id_t *> stream_infos);
void rtpAnalysisDialogRemoveRtpStreams(QVector<rtpstream_id_t *> stream_infos);
public slots:
void rtpAnalysisReplace();
void rtpAnalysisAdd();
void rtpAnalysisRemove();
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
Qt: Qt6.2 compatibility fixes Wireshark successfully compiles on Windows with Qt6.2 with following cmake options: -DUSE_qt6=ON -DDISABLE_ERROR=ON QCustomPlot QT 6.2 patch is taken from QCustomPlot forum post by miccs.
2021-11-28 11:43:27 +00:00
#endif
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
protected:
VoIP Dialogs: Refactoring of singleton windows to factory methods Singletons moved from main_window to each class's static open<NameOfClass> method: - RtpPlayerDialog - RtpStreamDialog - VoipCallsDialog - RtpAnalysisDialog Fixed issue with selecting RTP stream in sequence dialog. When user selected a stream and moved mouse to Rtp Player button and pressed it, incorrect RTP stream was sent to it.
2021-05-09 14:29:11 +00:00
explicit RtpPlayerDialog(QWidget &parent, CaptureFile &cf, bool capture_running);
Qt: Qt6.2 compatibility fixes Wireshark successfully compiles on Windows with Qt6.2 with following cmake options: -DUSE_qt6=ON -DDISABLE_ERROR=ON QCustomPlot QT 6.2 patch is taken from QCustomPlot forum post by miccs.
2021-11-28 11:43:27 +00:00
#ifdef QT_MULTIMEDIA_LIB
VoIP Dialogs: Refactoring of singleton windows to factory methods Singletons moved from main_window to each class's static open<NameOfClass> method: - RtpPlayerDialog - RtpStreamDialog - VoipCallsDialog - RtpAnalysisDialog Fixed issue with selecting RTP stream in sequence dialog. When user selected a stream and moved mouse to Rtp Player button and pressed it, incorrect RTP stream was sent to it.
2021-05-09 14:29:11 +00:00
~RtpPlayerDialog();
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
virtual void showEvent(QShowEvent *);
RTP dissector: Remove unused functions Patch removes (just comments out) functions not used by the code as found by Martin Mathieson.
2021-02-05 09:19:48 +00:00
void contextMenuEvent(QContextMenuEvent *event);
Rtp player: It is possible to select multiple rows in stream's list and graph Changes: - It is possible to select multiple streams in list and in graph - Select All/None/Invert implemented in list of streams and in graph - Indication of "Selected" stream redesigned in graph - Mouse hovering shows related row/wave - All operations adapted to multiselection
2021-03-09 00:58:15 +00:00
bool eventFilter(QObject *obj, QEvent *event);
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
private slots:
RTP Player: Dialog is nonmodal now and can be called multiple ways Changes: - refactored main_dialog handling of telephony dialogs - RTP Player dialog is nonmodal now and can be left open - it is possible to issue three actions on RTP Player dialog from other dialogs (other dialog have selected set of RTP streams before action) - replace - removes existing streams from RTP dialog and shows new set - add - adds new set to existing list in RTP dialog - remove - remove streams in set from list in RTP dialog - Sequence Dialog: - was modified to hold rtpstream_info_t for RTP streams - added Play button - VoIP features (RTP Play button, select/deselect RTP stream) are disabled after creation and must be enabled. It handles that RTP Play button is not shown e.g. in TCP sequence show
2021-03-27 20:01:43 +00:00
/** Retap the capture file, reading RTP packets that match the
Split RTP player tapping, decoding, and plotting. In RtpAudioStream split tapping+decoding into separate member functions. Store RTP payloads in memory. In RtpPlayerDialog split tapping+plotting. This more closely resembles what we're doing in the GTK+ UI and paves the way for jitter support and other changes. Change-Id: I244c225cec8930545622e6582b7be35ebe45b237 Reviewed-on: https://code.wireshark.org/review/11195 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-10-20 15:31:52 +00:00
* streams added using ::addRtpStream.
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
*/
Split RTP player tapping, decoding, and plotting. In RtpAudioStream split tapping+decoding into separate member functions. Store RTP payloads in memory. In RtpPlayerDialog split tapping+plotting. This more closely resembles what we're doing in the GTK+ UI and paves the way for jitter support and other changes. Change-Id: I244c225cec8930545622e6582b7be35ebe45b237 Reviewed-on: https://code.wireshark.org/review/11195 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-10-20 15:31:52 +00:00
void retapPackets();
RTP Player: Added button 'Refresh streams' for live capture
2021-04-22 19:33:11 +00:00
void captureEvent(CaptureEvent e);
Split RTP player tapping, decoding, and plotting. In RtpAudioStream split tapping+decoding into separate member functions. Store RTP payloads in memory. In RtpPlayerDialog split tapping+plotting. This more closely resembles what we're doing in the GTK+ UI and paves the way for jitter support and other changes. Change-Id: I244c225cec8930545622e6582b7be35ebe45b237 Reviewed-on: https://code.wireshark.org/review/11195 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-10-20 15:31:52 +00:00
/** Clear, decode, and redraw each stream.
*/
Add jitter logic to RtpAudioStream. Copy the jitter logic from rtp_player.c to rtp_audio_stream.cpp. This still isn't correct but the RTP player should now be complete enough to start looking at the bug list at the top of rtp_player_dialog.cpp. Disable timing and jitter controls while we're playing while we're here. Fixes bug 11635. Bug: 11635 Change-Id: Ie583ade522702cbe1bbcea4475a535caa1d74fa2 Reviewed-on: https://code.wireshark.org/review/11295 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-10-26 15:34:58 +00:00
void rescanPackets(bool rescale_axes = false);
RTP Player dialog improved Functional changes: Audio routing information is now stored in audio stream and not in table. It is handled by separate utils/rtp_audio_routing.cpp class which is able to convert it between mono/stereo etc. There is new utils/rtp_audio_routing_filter.cpp class which is able to route mono audio stream to any audio channel. Sample file separated from audio stream file - sample file is generated only during recap. So when we need new waveform, we just use existing sample file and no recap required - it is much faster. Audio stream exports just mono audio. Mono audio is then expanded to stereo and correct channel by AudioRoutingFilter during play as required. So when audio routing is changed, no recap and no audio export is required. When audio stream is muted, no audio is produced nor played. Most of signals between RtpPlayerDialog and RtpAudioStream were removed. Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams). Play possition is not received from every playing stream but from independent silence stream. Added Mute/Unmute function. Optimalization: When audio routing is changed, just graphs are updated. No retap nor audio decoding is required. When TOD is changed, just graphs are updated. No retap nor audio decoding is required.
2021-02-08 21:35:12 +00:00
void createPlot(bool rescale_axes = false);
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
void updateWidgets();
Rtp player: It is possible to select multiple rows in stream's list and graph Changes: - It is possible to select multiple streams in list and in graph - Select All/None/Invert implemented in list of streams and in graph - Indication of "Selected" stream redesigned in graph - Mouse hovering shows related row/wave - All operations adapted to multiselection
2021-03-09 00:58:15 +00:00
void itemEntered(QTreeWidgetItem *item, int column);
void mouseMovePlot(QMouseEvent *event);
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
void graphClicked(QMouseEvent *event);
rtp_player: Player is able to set start of audio play by double click Patch adds ability to set start of audio play by double clicking on waveform. Patch fixes unreported issue with placing waveform at incorrect place when switched relative/absolute time mode (check/uncheck Time of Day). Change-Id: Ib8ce24aea870e2443e033afbb6d6e9fbcf222431 Reviewed-on: https://code.wireshark.org/review/35621 Petri-Dish: Anders Broman <a.broman58@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-01 23:38:35 +00:00
void graphDoubleClicked(QMouseEvent *event);
RTP Player: Selected waveform is emphased. Waveform can be selected by click. When there is many waveforms, it is not clear which one is selected in list. It is important for selecting packets from graph. Patch emphasis selected waveform. In addition, it allows to select waveform by clicking on it.
2021-02-04 13:23:43 +00:00
void plotClicked(QCPAbstractPlottable *plottable, int dataIndex, QMouseEvent *event);
Qt: Make the RTP player output device selectable. Add a combobox for selecting the output device and populate it with our available devices. Let the user know if our output format isn't supported. Ping-Bug: 13105 Change-Id: I299c7d0f191bb66d93896338036000e2c377781f Reviewed-on: https://code.wireshark.org/review/19046 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Peter Wu <peter@lekensteyn.nl> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2016-12-02 23:52:02 +00:00
void updateHintLabel();
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
void resetXAxis();
RTP Player: UI improvements Changes: - all waveforms has common scale therefore louder/quiter signal is visible - when stream/streams are deleted from view, Y axis is rescaled and waveforms are rearranged to reuse empty space
2021-03-23 08:48:20 +00:00
void updateGraphs();
RTP Player: Parallel playlist operations avoided Changes: - UI is locked during adding new streams - When decoding of stream fails, it is reported in UI
2021-04-13 11:42:57 +00:00
void playFinished(RtpAudioStream *stream, QAudio::Error error);
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
void setPlayPosition(double secs);
RTP Player dialog improved Functional changes: Audio routing information is now stored in audio stream and not in table. It is handled by separate utils/rtp_audio_routing.cpp class which is able to convert it between mono/stereo etc. There is new utils/rtp_audio_routing_filter.cpp class which is able to route mono audio stream to any audio channel. Sample file separated from audio stream file - sample file is generated only during recap. So when we need new waveform, we just use existing sample file and no recap required - it is much faster. Audio stream exports just mono audio. Mono audio is then expanded to stereo and correct channel by AudioRoutingFilter during play as required. So when audio routing is changed, no recap and no audio export is required. When audio stream is muted, no audio is produced nor played. Most of signals between RtpPlayerDialog and RtpAudioStream were removed. Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams). Play possition is not received from every playing stream but from independent silence stream. Added Mute/Unmute function. Optimalization: When audio routing is changed, just graphs are updated. No retap nor audio decoding is required. When TOD is changed, just graphs are updated. No retap nor audio decoding is required.
2021-02-08 21:35:12 +00:00
void setPlaybackError(const QString playback_error);
void changeAudioRoutingOnItem(QTreeWidgetItem *ti, AudioRouting new_audio_routing);
void changeAudioRouting(AudioRouting new_audio_routing);
void invertAudioMutingOnItem(QTreeWidgetItem *ti);
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
void on_playButton_clicked();
RTP Player dialog improved Functional changes: Audio routing information is now stored in audio stream and not in table. It is handled by separate utils/rtp_audio_routing.cpp class which is able to convert it between mono/stereo etc. There is new utils/rtp_audio_routing_filter.cpp class which is able to route mono audio stream to any audio channel. Sample file separated from audio stream file - sample file is generated only during recap. So when we need new waveform, we just use existing sample file and no recap required - it is much faster. Audio stream exports just mono audio. Mono audio is then expanded to stereo and correct channel by AudioRoutingFilter during play as required. So when audio routing is changed, no recap and no audio export is required. When audio stream is muted, no audio is produced nor played. Most of signals between RtpPlayerDialog and RtpAudioStream were removed. Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams). Play possition is not received from every playing stream but from independent silence stream. Added Mute/Unmute function. Optimalization: When audio routing is changed, just graphs are updated. No retap nor audio decoding is required. When TOD is changed, just graphs are updated. No retap nor audio decoding is required.
2021-02-08 21:35:12 +00:00
void on_pauseButton_clicked();
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
void on_stopButton_clicked();
void on_actionReset_triggered();
void on_actionZoomIn_triggered();
void on_actionZoomOut_triggered();
void on_actionMoveLeft10_triggered();
void on_actionMoveRight10_triggered();
void on_actionMoveLeft1_triggered();
void on_actionMoveRight1_triggered();
void on_actionGoToPacket_triggered();
RTP player: It is possible to select SETUP packet related to RTP stream When SETUP (or related signalling packet) exists, it is selected by Shift+G. If RTP stream is "alone", first RTP packet of the stream is selected.
2021-03-09 13:47:07 +00:00
void on_actionGoToSetupPacketPlot_triggered();
void on_actionGoToSetupPacketTree_triggered();
RTP Player: Possibility to remove stream from RTP player When many streams are analyzed, some streams can be found as useless during replay. Patch adds option to remove it from list in RTP player. Once stream is removed, it can't be returned except close/reopen RTP player with new selection.
2021-02-04 14:03:45 +00:00
void on_actionRemoveStream_triggered();
RTP dissector: Remove unused functions Patch removes (just comments out) functions not used by the code as found by Martin Mathieson.
2021-02-05 09:19:48 +00:00
void on_actionAudioRoutingP_triggered();
void on_actionAudioRoutingL_triggered();
void on_actionAudioRoutingLR_triggered();
void on_actionAudioRoutingR_triggered();
RTP Player: Added additional shortcuts Added shortcuts: - Mute/Unmute/Invert Mute - Play/Pause/Stop Removed outdated help in tooltip text. It makes no sense to use it, all actions are available in context menu.
2021-03-11 08:22:15 +00:00
void on_actionAudioRoutingMute_triggered();
void on_actionAudioRoutingUnmute_triggered();
void on_actionAudioRoutingMuteInvert_triggered();
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
void on_streamTreeWidget_itemSelectionChanged();
rtp_player_dialog: Route audio for a stream to left/right speaker in RTP player Column 'Play' added to player. Double click on a stream in the column changes audio routing for the stream. When soundcard supports only one channel, there are Mute/Play option. When soundcard supports two or more channels, there are Mute/L/L+R/R options. Muted channel is drawn with dotted line. Change-Id: If120c902195da46f98a1663c589f20c6a1da0ba7 Reviewed-on: https://code.wireshark.org/review/35687 Petri-Dish: Alexis La Goutte <alexis.lagoutte@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-07 21:16:42 +00:00
void on_streamTreeWidget_itemDoubleClicked(QTreeWidgetItem *item, const int column);
Qt: Fix a couple of slot names. QComboBox emits currentTextChanged with a QString argument, not currentIndexChanged. Switch to new-style connections where we can.
2022-06-21 19:17:56 +00:00
void on_outputDeviceComboBox_currentTextChanged(const QString &);
void on_outputAudioRate_currentTextChanged(const QString &);
Add jitter logic to RtpAudioStream. Copy the jitter logic from rtp_player.c to rtp_audio_stream.cpp. This still isn't correct but the RTP player should now be complete enough to start looking at the bug list at the top of rtp_player_dialog.cpp. Disable timing and jitter controls while we're playing while we're here. Fixes bug 11635. Bug: 11635 Change-Id: Ie583ade522702cbe1bbcea4475a535caa1d74fa2 Reviewed-on: https://code.wireshark.org/review/11295 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-10-26 15:34:58 +00:00
void on_jitterSpinBox_valueChanged(double);
void on_timingComboBox_currentIndexChanged(int);
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
void on_todCheckBox_toggled(bool checked);
void on_buttonBox_helpRequested();
Rtp player: It is possible to select multiple rows in stream's list and graph Changes: - It is possible to select multiple streams in list and in graph - Select All/None/Invert implemented in list of streams and in graph - Indication of "Selected" stream redesigned in graph - Mouse hovering shows related row/wave - All operations adapted to multiselection
2021-03-09 00:58:15 +00:00
void on_actionSelectAll_triggered();
void on_actionSelectInvert_triggered();
void on_actionSelectNone_triggered();
RTP Player dialog improved Functional changes: Audio routing information is now stored in audio stream and not in table. It is handled by separate utils/rtp_audio_routing.cpp class which is able to convert it between mono/stereo etc. There is new utils/rtp_audio_routing_filter.cpp class which is able to route mono audio stream to any audio channel. Sample file separated from audio stream file - sample file is generated only during recap. So when we need new waveform, we just use existing sample file and no recap required - it is much faster. Audio stream exports just mono audio. Mono audio is then expanded to stereo and correct channel by AudioRoutingFilter during play as required. So when audio routing is changed, no recap and no audio export is required. When audio stream is muted, no audio is produced nor played. Most of signals between RtpPlayerDialog and RtpAudioStream were removed. Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams). Play possition is not received from every playing stream but from independent silence stream. Added Mute/Unmute function. Optimalization: When audio routing is changed, just graphs are updated. No retap nor audio decoding is required. When TOD is changed, just graphs are updated. No retap nor audio decoding is required.
2021-02-08 21:35:12 +00:00
void outputNotify();
RTP Player: Added additional shortcuts Added shortcuts: - Mute/Unmute/Invert Mute - Play/Pause/Stop Removed outdated help in tooltip text. It makes no sense to use it, all actions are available in context menu.
2021-03-11 08:22:15 +00:00
void on_actionPlay_triggered();
void on_actionStop_triggered();
RTP Player: Export of audio can start at position of play cursor Added additional option for audio exporting 'From cursor'. Documentation updated.
2021-04-27 20:01:16 +00:00
void on_actionSaveAudioFromCursor_triggered();
RTP Player: Added ability to save streams as .au and .wav Features: - saves multiple streams (all selected and unmuted) - saves streams same way they are played (jitter buffer, sampling, ...) - only streams with audio (play rate >0) are exported - streams with play rate == 0 are silently ignored even selected for export - all exported streams must use same play rate (user can change it before save)
2021-03-31 14:52:06 +00:00
void on_actionSaveAudioSyncStream_triggered();
void on_actionSaveAudioSyncFile_triggered();
RTP Player: Added ability to save streams as .au and .wav Features: - saves multiple streams (all selected and unmuted) - saves streams same way they are played (jitter buffer, sampling, ...) - only streams with audio (play rate >0) are exported - streams with play rate == 0 are silently ignored even selected for export - all exported streams must use same play rate (user can change it before save)
2021-04-01 08:07:22 +00:00
void on_actionSavePayload_triggered();
VoIP dialogs: Improvements and new functions Changes: - RTP Player added to Telephony/RTP menu. - When openning RTP Analysis or RTP Player from RTP menu, just selected stream is added. When Ctrl is hold during opening, reverse stream is searched and added too. - RTP Player: Added tool to select/deselect all inaudible streams - RTP Player: Added Prepare Filter button - RTP Player: Added Analyze button - RTP Analysis: Added Prepare Filter button - documentation updated Code changes: - RTP Player::rescanPacket() is not fired multiple times during rate change and during dialog creation - Error shown in RTP player is cleared after every new decode of streams - RTP Player handles case when Qt do not emit stop stream event - "Select" menu code unified between dialogs> - RTP Player: Audio routing menu unified - buttons are connected to actions by signals() - Analyze dialog is called by list of rtpstream_id, not rtpstream_info
2021-04-14 13:47:07 +00:00
void on_actionSelectInaudible_triggered();
void on_actionDeselectInaudible_triggered();
void on_actionPrepareFilter_triggered();
RTP Player: Added button 'Refresh streams' for live capture
2021-04-22 19:33:11 +00:00
void on_actionReadCapture_triggered();
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
Qt: Update the RTP stream UI to support Qt6Multimedia. Fixes #18115
2022-05-24 01:59:35 +00:00
#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
void sinkStateChanged();
#endif
Qt: Qt6.2 compatibility fixes Wireshark successfully compiles on Windows with Qt6.2 with following cmake options: -DUSE_qt6=ON -DDISABLE_ERROR=ON QCustomPlot QT 6.2 patch is taken from QCustomPlot forum post by miccs.
2021-11-28 11:43:27 +00:00
#endif
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
private:
VoIP Dialogs: Refactoring of singleton windows to factory methods Singletons moved from main_window to each class's static open<NameOfClass> method: - RtpPlayerDialog - RtpStreamDialog - VoipCallsDialog - RtpAnalysisDialog Fixed issue with selecting RTP stream in sequence dialog. When user selected a stream and moved mouse to Rtp Player button and pressed it, incorrect RTP stream was sent to it.
2021-05-09 14:29:11 +00:00
static RtpPlayerDialog *pinstance_;
Fix: Fixed issue with deadlock when same RTP dialog opened twice RTP dialogs can stay opened, therefore calls of its functions are protected by locks. There was issue that same mutex was used during construction of the dialog and calling functions. It created possible deadlock. Change separates lock used for dialog creation and lock for function calls. When function call lock is locked, new calls are ignored and warning is printed to STDERR. Showing a dialog with warning looks too intrusive to me. Fixes #18025
2022-04-04 19:21:26 +00:00
static std::mutex init_mutex_;
static std::mutex run_mutex_;
VoIP Dialogs: Refactoring of singleton windows to factory methods Singletons moved from main_window to each class's static open<NameOfClass> method: - RtpPlayerDialog - RtpStreamDialog - VoipCallsDialog - RtpAnalysisDialog Fixed issue with selecting RTP stream in sequence dialog. When user selected a stream and moved mouse to Rtp Player button and pressed it, incorrect RTP stream was sent to it.
2021-05-09 14:29:11 +00:00
Qt: Qt6.2 compatibility fixes Wireshark successfully compiles on Windows with Qt6.2 with following cmake options: -DUSE_qt6=ON -DDISABLE_ERROR=ON QCustomPlot QT 6.2 patch is taken from QCustomPlot forum post by miccs.
2021-11-28 11:43:27 +00:00
#ifdef QT_MULTIMEDIA_LIB
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
Ui::RtpPlayerDialog *ui;
RTP dissector: Remove unused functions Patch removes (just comments out) functions not used by the code as found by Martin Mathieson.
2021-02-05 09:19:48 +00:00
QMenu *graph_ctx_menu_;
QMenu *list_ctx_menu_;
rtp_player: Player is able to set start of audio play by double click Patch adds ability to set start of audio play by double clicking on waveform. Patch fixes unreported issue with placing waveform at incorrect place when switched relative/absolute time mode (check/uncheck Time of Day). Change-Id: Ib8ce24aea870e2443e033afbb6d6e9fbcf222431 Reviewed-on: https://code.wireshark.org/review/35621 Petri-Dish: Anders Broman <a.broman58@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-01 23:38:35 +00:00
double first_stream_rel_start_time_; // Relative start time of first stream
double first_stream_abs_start_time_; // Absolute start time of first stream
double first_stream_rel_stop_time_; // Relative end time of first stream (ued for streams_length_ calculation
double streams_length_; // Difference between start of first stream and end of last stream
double start_marker_time_; // Always relative time to start of the capture
RTP Player dialog improved Functional changes: Audio routing information is now stored in audio stream and not in table. It is handled by separate utils/rtp_audio_routing.cpp class which is able to convert it between mono/stereo etc. There is new utils/rtp_audio_routing_filter.cpp class which is able to route mono audio stream to any audio channel. Sample file separated from audio stream file - sample file is generated only during recap. So when we need new waveform, we just use existing sample file and no recap required - it is much faster. Audio stream exports just mono audio. Mono audio is then expanded to stereo and correct channel by AudioRoutingFilter during play as required. So when audio routing is changed, no recap and no audio export is required. When audio stream is muted, no audio is produced nor played. Most of signals between RtpPlayerDialog and RtpAudioStream were removed. Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams). Play possition is not received from every playing stream but from independent silence stream. Added Mute/Unmute function. Optimalization: When audio routing is changed, just graphs are updated. No retap nor audio decoding is required. When TOD is changed, just graphs are updated. No retap nor audio decoding is required.
2021-02-08 21:35:12 +00:00
double start_marker_time_play_; // Copy when play started
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
QCPItemStraightLine *cur_play_pos_;
rtp_player: Player is able to set start of audio play by double click Patch adds ability to set start of audio play by double clicking on waveform. Patch fixes unreported issue with placing waveform at incorrect place when switched relative/absolute time mode (check/uncheck Time of Day). Change-Id: Ib8ce24aea870e2443e033afbb6d6e9fbcf222431 Reviewed-on: https://code.wireshark.org/review/35621 Petri-Dish: Anders Broman <a.broman58@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-01 23:38:35 +00:00
QCPItemStraightLine *start_marker_pos_;
Qt: Make the RTP player output device selectable. Add a combobox for selecting the output device and populate it with our available devices. Let the user know if our output format isn't supported. Ping-Bug: 13105 Change-Id: I299c7d0f191bb66d93896338036000e2c377781f Reviewed-on: https://code.wireshark.org/review/19046 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Peter Wu <peter@lekensteyn.nl> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2016-12-02 23:52:02 +00:00
QString playback_error_;
Update to qcustomplot 2.0.1 Upgrade the internal version of QCustomPlot to 2.0.1 Change-Id: I1eb372d8e6a2f6c1bbdde4c74596785bf2d405c2 Reviewed-on: https://code.wireshark.org/review/17980 Reviewed-by: Roland Knall <rknall@gmail.com> Petri-Dish: Roland Knall <rknall@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2016-09-25 09:57:55 +00:00
QSharedPointer<QCPAxisTicker> number_ticker_;
QSharedPointer<QCPAxisTickerDateTime> datetime_ticker_;
rtp_player_dialog: Route audio for a stream to left/right speaker in RTP player Column 'Play' added to player. Double click on a stream in the column changes audio routing for the stream. When soundcard supports only one channel, there are Mute/Play option. When soundcard supports two or more channels, there are Mute/L/L+R/R options. Muted channel is drawn with dotted line. Change-Id: If120c902195da46f98a1663c589f20c6a1da0ba7 Reviewed-on: https://code.wireshark.org/review/35687 Petri-Dish: Alexis La Goutte <alexis.lagoutte@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-07 21:16:42 +00:00
bool stereo_available_;
RTP Player dialog improved Functional changes: Audio routing information is now stored in audio stream and not in table. It is handled by separate utils/rtp_audio_routing.cpp class which is able to convert it between mono/stereo etc. There is new utils/rtp_audio_routing_filter.cpp class which is able to route mono audio stream to any audio channel. Sample file separated from audio stream file - sample file is generated only during recap. So when we need new waveform, we just use existing sample file and no recap required - it is much faster. Audio stream exports just mono audio. Mono audio is then expanded to stereo and correct channel by AudioRoutingFilter during play as required. So when audio routing is changed, no recap and no audio export is required. When audio stream is muted, no audio is produced nor played. Most of signals between RtpPlayerDialog and RtpAudioStream were removed. Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams). Play possition is not received from every playing stream but from independent silence stream. Added Mute/Unmute function. Optimalization: When audio routing is changed, just graphs are updated. No retap nor audio decoding is required. When TOD is changed, just graphs are updated. No retap nor audio decoding is required.
2021-02-08 21:35:12 +00:00
QList<RtpAudioStream *> playing_streams_;
Qt: Update the RTP stream UI to support Qt6Multimedia. Fixes #18115
2022-05-24 01:59:35 +00:00
#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
QAudioSink *marker_stream_;
QTimer notify_timer_;
Qt6: Audio code review Changes: - audio rates are fixed, I selected common ones - fixed signal names - they didn't matched expected autogenerated ones - first play notification delay on Qt6 compensated
2022-07-08 13:10:21 +00:00
qint64 notify_timer_start_diff_; // Used to shift play cursor to correct place
Qt: Update the RTP stream UI to support Qt6Multimedia. Fixes #18115
2022-05-24 01:59:35 +00:00
#else
RTP Player dialog improved Functional changes: Audio routing information is now stored in audio stream and not in table. It is handled by separate utils/rtp_audio_routing.cpp class which is able to convert it between mono/stereo etc. There is new utils/rtp_audio_routing_filter.cpp class which is able to route mono audio stream to any audio channel. Sample file separated from audio stream file - sample file is generated only during recap. So when we need new waveform, we just use existing sample file and no recap required - it is much faster. Audio stream exports just mono audio. Mono audio is then expanded to stereo and correct channel by AudioRoutingFilter during play as required. So when audio routing is changed, no recap and no audio export is required. When audio stream is muted, no audio is produced nor played. Most of signals between RtpPlayerDialog and RtpAudioStream were removed. Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams). Play possition is not received from every playing stream but from independent silence stream. Added Mute/Unmute function. Optimalization: When audio routing is changed, just graphs are updated. No retap nor audio decoding is required. When TOD is changed, just graphs are updated. No retap nor audio decoding is required.
2021-02-08 21:35:12 +00:00
QAudioOutput *marker_stream_;
Qt: Update the RTP stream UI to support Qt6Multimedia. Fixes #18115
2022-05-24 01:59:35 +00:00
#endif
RTP Player: Fixed issue with hanging the player at end of play on Windows Changes: - Fixed issue with hanging the player. The issue is in Qt - internal Mutex is locked when Qt calls outputStateChanged() and you can't call any other action on the audio object - Fixed issue when play marker stream was running forever - Removed !1855 because it introduces delay on play on Windows platform
2021-04-08 09:15:08 +00:00
quint32 marker_stream_requested_out_rate_;
Rtp player: It is possible to select multiple rows in stream's list and graph Changes: - It is possible to select multiple streams in list and in graph - Select All/None/Invert implemented in list of streams and in graph - Indication of "Selected" stream redesigned in graph - Mouse hovering shows related row/wave - All operations adapted to multiselection
2021-03-09 00:58:15 +00:00
QTreeWidgetItem *last_ti_;
RTP Player: Fix for #16452 Reason for #16452 is that RTP Player dialog is destroyed before tap finishes when Escape key pressed. The only solution I found is to check whether dialog does exists when tap finishes. If so, data are processed and shown. If dialog was destroyed in meanwhile, no data processing is done.
2021-01-02 00:58:21 +00:00
bool listener_removed_;
RTP Player: Added button 'Refresh streams' for live capture
2021-04-22 19:33:11 +00:00
QPushButton *read_btn_;
VoIP dialogs: Default actions for buttons works as before When button is pressed or triggered by shortcut, it opens same window as before. User can click small arrow next to button and it open menu with all new actions e.g. Set/Add/Remove for RTP Player. Documentation updated.
2021-05-01 22:33:54 +00:00
QToolButton *inaudible_btn_;
QToolButton *analyze_btn_;
VoIP dialogs: Improvements and new functions Changes: - RTP Player added to Telephony/RTP menu. - When openning RTP Analysis or RTP Player from RTP menu, just selected stream is added. When Ctrl is hold during opening, reverse stream is searched and added too. - RTP Player: Added tool to select/deselect all inaudible streams - RTP Player: Added Prepare Filter button - RTP Player: Added Analyze button - RTP Analysis: Added Prepare Filter button - documentation updated Code changes: - RTP Player::rescanPacket() is not fired multiple times during rate change and during dialog creation - Error shown in RTP player is cleared after every new decode of streams - RTP Player handles case when Qt do not emit stop stream event - "Select" menu code unified between dialogs> - RTP Player: Audio routing menu unified - buttons are connected to actions by signals() - Analyze dialog is called by list of rtpstream_id, not rtpstream_info
2021-04-14 13:47:07 +00:00
QPushButton *prepare_btn_;
RTP Player: UI improvements
2021-04-06 11:23:00 +00:00
QPushButton *export_btn_;
VoIP dialogs: Performance improvements Retap and UI response are much faster when many RTP streams are processed. RTP Streams/Analyse 1000+, RTP Player 500+. Changes: - RTP streams are searched with hash, not by iterating over list. - UI operations do not redraw screen after every change, just after all changes. UI is locked when rereading packets. - Sample list during RTP decoding is stored in memory so wireshark uses just half of opened files for audio decoding than before. - Analysis window checkbox area is limited in height - Dialogs shows shows count of streams, count of selected streams and count of unmuted streams - Documentation extended with chapter about RTP decoding parameters - Documentation extended with performance estimates
2021-04-13 14:38:13 +00:00
QMultiHash<guint, RtpAudioStream *> stream_hash_;
bool block_redraw_;
int lock_ui_;
RTP Player: Added button 'Refresh streams' for live capture
2021-04-22 19:33:11 +00:00
bool read_capture_enabled_;
RTP Player: Player is able to skip silence during playback Code is NOT able to do VAD (Voice Activity Detection) so audio silence (sequence of equal samples) nor noise are not recognized as silence. Just missing RTP (Confort Noise, interupted RTP, ...) and muted streams are recognized as silence for this feature. User can control duration of shortest silence to skip. Updated documentation.
2021-05-01 11:51:31 +00:00
double silence_skipped_time_;
RTP Player: Fix for #16452 Reason for #16452 is that RTP Player dialog is destroyed before tap finishes when Escape key pressed. The only solution I found is to check whether dialog does exists when tap finishes. If so, data are processed and shown. If dialog was destroyed in meanwhile, no data processing is done.
2021-01-02 00:58:21 +00:00
RTP: Common functions for allocation/deallocation of rtpstream_info_t Change-Id: I9a0a11d238473a7c57d85547dca0713ed421a500 Reviewed-on: https://code.wireshark.org/review/28417 Petri-Dish: Alexis La Goutte <alexis.lagoutte@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2018-06-21 13:16:03 +00:00
// const QString streamKey(const rtpstream_info_t *rtpstream);
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
// const QString streamKey(const packet_info *pinfo, const struct _rtp_info *rtpinfo);
// Tap callbacks
// static void tapReset(void *tapinfo_ptr);
tap: Adding flags for tap_packet This allows flags to be passed by the registering listener to the collection of information
2022-06-09 15:47:35 +00:00
static tap_packet_status tapPacket(void *tapinfo_ptr, packet_info *pinfo, epan_dissect_t *, const void *rtpinfo_ptr, tap_flags_t flags);
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
static void tapDraw(void *tapinfo_ptr);
void addPacket(packet_info *pinfo, const struct _rtp_info *rtpinfo);
void zoomXAxis(bool in);
void panXAxis(int x_pixels);
rtp_player_dialog: Route audio for a stream to left/right speaker in RTP player Column 'Play' added to player. Double click on a stream in the column changes audio routing for the stream. When soundcard supports only one channel, there are Mute/Play option. When soundcard supports two or more channels, there are Mute/L/L+R/R options. Muted channel is drawn with dotted line. Change-Id: If120c902195da46f98a1663c589f20c6a1da0ba7 Reviewed-on: https://code.wireshark.org/review/35687 Petri-Dish: Alexis La Goutte <alexis.lagoutte@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-07 21:16:42 +00:00
const QString getFormatedTime(double f_time);
rtp_player: Player is able to set start of audio play by double click Patch adds ability to set start of audio play by double clicking on waveform. Patch fixes unreported issue with placing waveform at incorrect place when switched relative/absolute time mode (check/uncheck Time of Day). Change-Id: Ib8ce24aea870e2443e033afbb6d6e9fbcf222431 Reviewed-on: https://code.wireshark.org/review/35621 Petri-Dish: Anders Broman <a.broman58@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-01 23:38:35 +00:00
const QString getFormatedHoveredTime();
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
int getHoveredPacket();
Qt: Fixup the currentOutputDeviceName Q_PROPERTY. The CONSTANT attribute indicates that the same value will be returned every time. That isn't the case here so remove it. Change-Id: Ie7451e6aabcb4fa1a6960762d96ad190f32b3d7a Reviewed-on: https://code.wireshark.org/review/19130 Reviewed-by: Gerald Combs <gerald@wireshark.org> Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Michael Mann <mmann78@netscape.net>
2016-12-07 17:58:28 +00:00
QString currentOutputDeviceName();
rtp_player: Player is able to set start of audio play by double click Patch adds ability to set start of audio play by double clicking on waveform. Patch fixes unreported issue with placing waveform at incorrect place when switched relative/absolute time mode (check/uncheck Time of Day). Change-Id: Ib8ce24aea870e2443e033afbb6d6e9fbcf222431 Reviewed-on: https://code.wireshark.org/review/35621 Petri-Dish: Anders Broman <a.broman58@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-01 23:38:35 +00:00
double getStartPlayMarker();
void drawStartPlayMarker();
rtp_player_dialog: Route audio for a stream to left/right speaker in RTP player Column 'Play' added to player. Double click on a stream in the column changes audio routing for the stream. When soundcard supports only one channel, there are Mute/Play option. When soundcard supports two or more channels, there are Mute/L/L+R/R options. Muted channel is drawn with dotted line. Change-Id: If120c902195da46f98a1663c589f20c6a1da0ba7 Reviewed-on: https://code.wireshark.org/review/35687 Petri-Dish: Alexis La Goutte <alexis.lagoutte@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-07 21:16:42 +00:00
void setStartPlayMarker(double new_time);
RTP Player: Added button 'Refresh streams' for live capture
2021-04-22 19:33:11 +00:00
void updateStartStopTime(rtpstream_info_t *rtpstream, bool is_first);
RTP Player dialog improved Functional changes: Audio routing information is now stored in audio stream and not in table. It is handled by separate utils/rtp_audio_routing.cpp class which is able to convert it between mono/stereo etc. There is new utils/rtp_audio_routing_filter.cpp class which is able to route mono audio stream to any audio channel. Sample file separated from audio stream file - sample file is generated only during recap. So when we need new waveform, we just use existing sample file and no recap required - it is much faster. Audio stream exports just mono audio. Mono audio is then expanded to stereo and correct channel by AudioRoutingFilter during play as required. So when audio routing is changed, no recap and no audio export is required. When audio stream is muted, no audio is produced nor played. Most of signals between RtpPlayerDialog and RtpAudioStream were removed. Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams). Play possition is not received from every playing stream but from independent silence stream. Added Mute/Unmute function. Optimalization: When audio routing is changed, just graphs are updated. No retap nor audio decoding is required. When TOD is changed, just graphs are updated. No retap nor audio decoding is required.
2021-02-08 21:35:12 +00:00
void formatAudioRouting(QTreeWidgetItem *ti, AudioRouting audio_routing);
rtp_player_dialog: Route audio for a stream to left/right speaker in RTP player Column 'Play' added to player. Double click on a stream in the column changes audio routing for the stream. When soundcard supports only one channel, there are Mute/Play option. When soundcard supports two or more channels, there are Mute/L/L+R/R options. Muted channel is drawn with dotted line. Change-Id: If120c902195da46f98a1663c589f20c6a1da0ba7 Reviewed-on: https://code.wireshark.org/review/35687 Petri-Dish: Alexis La Goutte <alexis.lagoutte@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
2020-01-07 21:16:42 +00:00
bool isStereoAvailable();
Qt: Update the RTP stream UI to support Qt6Multimedia. Fixes #18115
2022-05-24 01:59:35 +00:00
#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
QAudioSink *getSilenceAudioOutput();
QAudioDevice getCurrentDeviceInfo();
#else
RTP Player dialog improved Functional changes: Audio routing information is now stored in audio stream and not in table. It is handled by separate utils/rtp_audio_routing.cpp class which is able to convert it between mono/stereo etc. There is new utils/rtp_audio_routing_filter.cpp class which is able to route mono audio stream to any audio channel. Sample file separated from audio stream file - sample file is generated only during recap. So when we need new waveform, we just use existing sample file and no recap required - it is much faster. Audio stream exports just mono audio. Mono audio is then expanded to stereo and correct channel by AudioRoutingFilter during play as required. So when audio routing is changed, no recap and no audio export is required. When audio stream is muted, no audio is produced nor played. Most of signals between RtpPlayerDialog and RtpAudioStream were removed. Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams). Play possition is not received from every playing stream but from independent silence stream. Added Mute/Unmute function. Optimalization: When audio routing is changed, just graphs are updated. No retap nor audio decoding is required. When TOD is changed, just graphs are updated. No retap nor audio decoding is required.
2021-02-08 21:35:12 +00:00
QAudioOutput *getSilenceAudioOutput();
QAudioDeviceInfo getCurrentDeviceInfo();
Qt: Update the RTP stream UI to support Qt6Multimedia. Fixes #18115
2022-05-24 01:59:35 +00:00
#endif
Rtp player: It is possible to select multiple rows in stream's list and graph Changes: - It is possible to select multiple streams in list and in graph - Select All/None/Invert implemented in list of streams and in graph - Indication of "Selected" stream redesigned in graph - Mouse hovering shows related row/wave - All operations adapted to multiselection
2021-03-09 00:58:15 +00:00
QTreeWidgetItem *findItemByCoords(QPoint point);
QTreeWidgetItem *findItem(QCPAbstractPlottable *plottable);
void handleItemHighlight(QTreeWidgetItem *ti, bool scroll);
void highlightItem(QTreeWidgetItem *ti, bool highlight);
void invertSelection();
RTP player: It is possible to select SETUP packet related to RTP stream When SETUP (or related signalling packet) exists, it is selected by Shift+G. If RTP stream is "alone", first RTP packet of the stream is selected.
2021-03-09 13:47:07 +00:00
void handleGoToSetupPacket(QTreeWidgetItem *ti);
RTP Player: Added button 'Refresh streams' for live capture
2021-04-22 19:33:11 +00:00
void addSingleRtpStream(rtpstream_id_t *id);
RTP Player: Dialog is nonmodal now and can be called multiple ways Changes: - refactored main_dialog handling of telephony dialogs - RTP Player dialog is nonmodal now and can be left open - it is possible to issue three actions on RTP Player dialog from other dialogs (other dialog have selected set of RTP streams before action) - replace - removes existing streams from RTP dialog and shows new set - add - adds new set to existing list in RTP dialog - remove - remove streams in set from list in RTP dialog - Sequence Dialog: - was modified to hold rtpstream_info_t for RTP streams - added Play button - VoIP features (RTP Play button, select/deselect RTP stream) are disabled after creation and must be enabled. It handles that RTP Play button is not shown e.g. in TCP sequence show
2021-03-27 20:01:43 +00:00
void removeRow(QTreeWidgetItem *ti);
RTP Player: Allow a user to select output rate Tool allows a user to replay at specific rate when there is any issue with autodetected rate by payloads. Offered rates are provided by selected audio device.
2021-03-29 09:37:21 +00:00
void fillAudioRateMenu();
RTP Player: Fixed issue with hanging the player at end of play on Windows Changes: - Fixed issue with hanging the player. The issue is in Qt - internal Mutex is locked when Qt calls outputStateChanged() and you can't call any other action on the audio object - Fixed issue when play marker stream was running forever - Removed !1855 because it introduces delay on play on Windows platform
2021-04-08 09:15:08 +00:00
void cleanupMarkerStream();
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
Qt: Update the RTP stream UI to support Qt6Multimedia. Fixes #18115
2022-05-24 01:59:35 +00:00
qint64 saveAudioHeaderAU(QFile *save_file, quint32 channels, unsigned audio_rate);
qint64 saveAudioHeaderWAV(QFile *save_file, quint32 channels, unsigned audio_rate, qint64 samples);
RTP Player: Memory consumption improvements Audio for play is now decoded and stored without silence parts. Changes: - ui/qt/utils/rtp_audio_file.cpp created to handle silence skipping - ui/qt/rtp_audio_stream.cpp refactored to support it - Fixed issue with exporting streams: File synchronized export was missing leading silence. - No line is shown in waveform graph if there is silence
2021-04-24 16:08:21 +00:00
bool writeAudioSilenceSamples(QFile *out_file, qint64 samples, int stream_count);
RTP Player: Added ability to save streams as .au and .wav Features: - saves multiple streams (all selected and unmuted) - saves streams same way they are played (jitter buffer, sampling, ...) - only streams with audio (play rate >0) are exported - streams with play rate == 0 are silently ignored even selected for export - all exported streams must use same play rate (user can change it before save)
2021-04-01 08:07:22 +00:00
bool writeAudioStreamsSamples(QFile *out_file, QVector<RtpAudioStream *> streams, bool swap_bytes);
save_audio_t selectFileAudioFormatAndName(QString *file_path);
save_payload_t selectFilePayloadFormatAndName(QString *file_path);
VoIP dialogs: Improvements and new functions Changes: - RTP Player added to Telephony/RTP menu. - When openning RTP Analysis or RTP Player from RTP menu, just selected stream is added. When Ctrl is hold during opening, reverse stream is searched and added too. - RTP Player: Added tool to select/deselect all inaudible streams - RTP Player: Added Prepare Filter button - RTP Player: Added Analyze button - RTP Analysis: Added Prepare Filter button - documentation updated Code changes: - RTP Player::rescanPacket() is not fired multiple times during rate change and during dialog creation - Error shown in RTP player is cleared after every new decode of streams - RTP Player handles case when Qt do not emit stop stream event - "Select" menu code unified between dialogs> - RTP Player: Audio routing menu unified - buttons are connected to actions by signals() - Analyze dialog is called by list of rtpstream_id, not rtpstream_info
2021-04-14 13:47:07 +00:00
QVector<RtpAudioStream *>getSelectedAudibleNonmutedAudioStreams();
RTP Player: Export of audio can start at position of play cursor Added additional option for audio exporting 'From cursor'. Documentation updated.
2021-04-27 20:01:16 +00:00
void saveAudio(save_mode_t save_mode);
RTP Player: Added ability to save streams as .au and .wav Features: - saves multiple streams (all selected and unmuted) - saves streams same way they are played (jitter buffer, sampling, ...) - only streams with audio (play rate >0) are exported - streams with play rate == 0 are silently ignored even selected for export - all exported streams must use same play rate (user can change it before save)
2021-04-01 08:07:22 +00:00
void savePayload();
RTP Player: Parallel playlist operations avoided Changes: - UI is locked during adding new streams - When decoding of stream fails, it is reported in UI
2021-04-13 11:42:57 +00:00
void lockUI();
void unlockUI();
VoIP dialogs: Improvements and new functions Changes: - RTP Player added to Telephony/RTP menu. - When openning RTP Analysis or RTP Player from RTP menu, just selected stream is added. When Ctrl is hold during opening, reverse stream is searched and added too. - RTP Player: Added tool to select/deselect all inaudible streams - RTP Player: Added Prepare Filter button - RTP Player: Added Analyze button - RTP Analysis: Added Prepare Filter button - documentation updated Code changes: - RTP Player::rescanPacket() is not fired multiple times during rate change and during dialog creation - Error shown in RTP player is cleared after every new decode of streams - RTP Player handles case when Qt do not emit stop stream event - "Select" menu code unified between dialogs> - RTP Player: Audio routing menu unified - buttons are connected to actions by signals() - Analyze dialog is called by list of rtpstream_id, not rtpstream_info
2021-04-14 13:47:07 +00:00
void selectInaudible(bool select);
QVector<rtpstream_id_t *>getSelectedRtpStreamIDs();
RTP Player: Added button 'Refresh streams' for live capture
2021-04-22 19:33:11 +00:00
void fillTappedColumns();
RTP Player: Added ability to save streams as .au and .wav Features: - saves multiple streams (all selected and unmuted) - saves streams same way they are played (jitter buffer, sampling, ...) - only streams with audio (play rate >0) are exported - streams with play rate == 0 are silently ignored even selected for export - all exported streams must use same play rate (user can change it before save)
2021-03-31 14:52:06 +00:00
Fix warnings introduced by "Qt: Initial RTP playback" Change-Id: I28ae077be535f32ef81ac370d6782033f219017d Reviewed-on: https://code.wireshark.org/review/10777 Reviewed-by: Peter Wu <peter@lekensteyn.nl> Petri-Dish: Peter Wu <peter@lekensteyn.nl> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Anders Broman <a.broman58@gmail.com> Reviewed-by: Michael Mann <mmann78@netscape.net>
2015-10-04 09:52:26 +00:00
#else // QT_MULTIMEDIA_LIB
Qt: Initial RTP playback. Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-13 00:51:40 +00:00
private:
Ui::RtpPlayerDialog *ui;
#endif // QT_MULTIMEDIA_LIB
};
#endif // RTP_PLAYER_DIALOG_H