Wireshark  4.3.0
The Wireshark network protocol analyzer
rtp_audio_stream.h
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1 
10 #ifndef RTPAUDIOSTREAM_H
11 #define RTPAUDIOSTREAM_H
12 
13 #include "config.h"
14 
15 #ifdef QT_MULTIMEDIA_LIB
16 
17 #include <epan/address.h>
18 #include <ui/rtp_stream.h>
21 #include <ui/rtp_media.h>
22 
23 #include <QAudio>
24 #include <QColor>
25 #include <QMap>
26 #include <QObject>
27 #include <QSet>
28 #include <QVector>
29 #include <QIODevice>
30 #include <QAudioOutput>
31 
32 class QAudioFormat;
33 #if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
34 class QAudioSink;
35 #else
36 class QAudioOutput;
37 #endif
38 class QIODevice;
39 
40 
41 class RtpAudioStream : public QObject
42 {
43  Q_OBJECT
44 public:
45  enum TimingMode { JitterBuffer, RtpTimestamp, Uninterrupted };
46 
47  explicit RtpAudioStream(QObject *parent, rtpstream_id_t *id, bool stereo_required);
48  ~RtpAudioStream();
49  bool isMatch(const rtpstream_id_t *id) const;
50  bool isMatch(const struct _packet_info *pinfo, const struct _rtp_info *rtp_info) const;
51  void addRtpPacket(const struct _packet_info *pinfo, const struct _rtp_info *rtp_info);
52  void clearPackets();
53  void reset(double global_start_time);
54  AudioRouting getAudioRouting();
55  void setAudioRouting(AudioRouting audio_routing);
56 #if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
57  void decode(QAudioDevice out_device);
58 #else
59  void decode(QAudioDeviceInfo out_device);
60 #endif
61 
62  double startRelTime() const { return start_rel_time_; }
63  double stopRelTime() const { return stop_rel_time_; }
64  unsigned sampleRate() const { return first_sample_rate_; }
65  unsigned playRate() const { return audio_out_rate_; }
66  void setRequestedPlayRate(unsigned new_rate) { audio_requested_out_rate_ = new_rate; }
67  const QStringList payloadNames() const;
68 
73  const QVector<double> visualTimestamps(bool relative = true);
80  const QVector<double> visualSamples(int y_offset = 0);
81 
86  const QVector<double> outOfSequenceTimestamps(bool relative = true);
87  int outOfSequence() { return static_cast<int>(out_of_seq_timestamps_.size()); }
93  const QVector<double> outOfSequenceSamples(int y_offset = 0);
94 
99  const QVector<double> jitterDroppedTimestamps(bool relative = true);
100  int jitterDropped() { return static_cast<int>(jitter_drop_timestamps_.size()); }
106  const QVector<double> jitterDroppedSamples(int y_offset = 0);
107 
112  const QVector<double> wrongTimestampTimestamps(bool relative = true);
113  int wrongTimestamps() { return static_cast<int>(wrong_timestamp_timestamps_.size()); }
119  const QVector<double> wrongTimestampSamples(int y_offset = 0);
120 
125  const QVector<double> insertedSilenceTimestamps(bool relative = true);
126  int insertedSilences() { return static_cast<int>(silence_timestamps_.size()); }
132  const QVector<double> insertedSilenceSamples(int y_offset = 0);
133 
134  quint32 nearestPacket(double timestamp, bool is_relative = true);
135 
136  QRgb color() { return color_; }
137  void setColor(QRgb color) { color_ = color; }
138 
139  QAudio::State outputState() const;
140 
141  void setJitterBufferSize(int jitter_buffer_size) { jitter_buffer_size_ = jitter_buffer_size; }
142  void setTimingMode(TimingMode timing_mode) { timing_mode_ = timing_mode; }
143  void setStartPlayTime(double start_play_time) { start_play_time_ = start_play_time; }
144 #if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
145  bool prepareForPlay(QAudioDevice out_device);
146 #else
147  bool prepareForPlay(QAudioDeviceInfo out_device);
148 #endif
149  void startPlaying();
150  void pausePlaying();
151  void stopPlaying();
152  void seekPlaying(qint64 samples);
153  void setStereoRequired(bool stereo_required) { stereo_required_ = stereo_required; }
154  qint16 getMaxSampleValue() { return max_sample_val_; }
155  void setMaxSampleValue(int16_t max_sample_val) { max_sample_val_used_ = max_sample_val; }
156  void seekSample(qint64 samples);
157  qint64 readSample(SAMPLE *sample);
158  qint64 getLeadSilenceSamples() { return prepend_samples_; }
159  qint64 getTotalSamples() { return (audio_file_->getTotalSamples()); }
160  qint64 getEndOfSilenceSample() { return (audio_file_->getEndOfSilenceSample()); }
161  double getEndOfSilenceTime() { return (double)getEndOfSilenceSample() / (double)playRate(); }
162  qint64 convertTimeToSamples(double time) { return (qint64)(time * playRate()); }
163  bool savePayload(QIODevice *file);
164  unsigned getHash() { return rtpstream_id_to_hash(&(id_)); }
165  rtpstream_id_t *getID() { return &(id_); }
166  QString getIDAsQString();
167  rtpstream_info_t *getStreamInfo() { return &rtpstream_; }
168 
169 signals:
170  void processedSecs(double secs);
171  void playbackError(const QString error_msg);
172  void finishedPlaying(RtpAudioStream *stream, QAudio::Error error);
173 
174 private:
175  // Used to identify unique streams.
176  // The GTK+ UI also uses the call number + current channel.
177  rtpstream_id_t id_;
178  rtpstream_info_t rtpstream_;
179  bool first_packet_;
180 
181  QVector<struct _rtp_packet *>rtp_packets_;
182  RtpAudioFile *audio_file_; // Stores waveform samples in sparse file
183  QIODevice *temp_file_;
184  struct _GHashTable *decoders_hash_;
185  double global_start_rel_time_;
186  double start_abs_offset_;
187  double start_rel_time_;
188  double stop_rel_time_;
189  qint64 prepend_samples_; // Count of silence samples at begin of the stream to align with other streams
190  AudioRouting audio_routing_;
191  bool stereo_required_;
192  quint32 first_sample_rate_;
193  quint32 audio_out_rate_;
194  quint32 audio_requested_out_rate_;
195  QSet<QString> payload_names_;
196  struct SpeexResamplerState_ *visual_resampler_;
197  QMap<double, quint32> packet_timestamps_;
198  QVector<qint16> visual_samples_;
199  QVector<double> out_of_seq_timestamps_;
200  QVector<double> jitter_drop_timestamps_;
201  QVector<double> wrong_timestamp_timestamps_;
202  QVector<double> silence_timestamps_;
203  qint16 max_sample_val_;
204  qint16 max_sample_val_used_;
205  QRgb color_;
206 
207  int jitter_buffer_size_;
208  TimingMode timing_mode_;
209  double start_play_time_;
210 
211  const QString formatDescription(const QAudioFormat & format);
212  QString currentOutputDevice();
213 
214 #if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
215  QAudioSink *audio_output_;
216  void decodeAudio(QAudioDevice out_device);
217  quint32 calculateAudioOutRate(QAudioDevice out_device, unsigned int sample_rate, unsigned int requested_out_rate);
218 #else
219  QAudioOutput *audio_output_;
220  void decodeAudio(QAudioDeviceInfo out_device);
221  quint32 calculateAudioOutRate(QAudioDeviceInfo out_device, unsigned int sample_rate, unsigned int requested_out_rate);
222 #endif
223  void decodeVisual();
224  SAMPLE *resizeBufferIfNeeded(SAMPLE *buff, int32_t *buff_bytes, qint64 requested_size);
225 
226 private slots:
227  void outputStateChanged(QAudio::State new_state);
228  void delayedStopStream();
229 };
230 
231 #endif // QT_MULTIMEDIA_LIB
232 
233 #endif // RTPAUDIOSTREAM_H
Definition: rtp_audio_routing.h:28
Definition: rtp_audio_file.h:42
unsigned rtpstream_id_to_hash(const rtpstream_id_t *id)
Definition: rtp_stream_id.c:89
Definition: packet_info.h:44
Definition: packet-rtp.h:29
Definition: rtp_stream_id.h:33
Definition: rtp_stream.h:40
Definition: stream.c:41