237 lines
8.8 KiB
C++
237 lines
8.8 KiB
C++
/** @file
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*
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* Wireshark - Network traffic analyzer
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* By Gerald Combs <gerald@wireshark.org>
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* Copyright 1998 Gerald Combs
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*
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* SPDX-License-Identifier: GPL-2.0-or-later
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*/
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#ifndef RTPAUDIOSTREAM_H
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#define RTPAUDIOSTREAM_H
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#include "config.h"
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#ifdef QT_MULTIMEDIA_LIB
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#include <glib.h>
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#include <epan/address.h>
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#include <ui/rtp_stream.h>
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#include <ui/qt/utils/rtp_audio_routing.h>
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#include <ui/qt/utils/rtp_audio_file.h>
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#include <ui/rtp_media.h>
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#include <QAudio>
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#include <QColor>
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#include <QMap>
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#include <QObject>
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#include <QSet>
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#include <QVector>
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#include <QIODevice>
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#include <QAudioOutput>
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class QAudioFormat;
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#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
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class QAudioSink;
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#else
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class QAudioOutput;
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#endif
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class QIODevice;
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class RtpAudioStream : public QObject
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{
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Q_OBJECT
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public:
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enum TimingMode { JitterBuffer, RtpTimestamp, Uninterrupted };
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explicit RtpAudioStream(QObject *parent, rtpstream_id_t *id, bool stereo_required);
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~RtpAudioStream();
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bool isMatch(const rtpstream_id_t *id) const;
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bool isMatch(const struct _packet_info *pinfo, const struct _rtp_info *rtp_info) const;
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void addRtpPacket(const struct _packet_info *pinfo, const struct _rtp_info *rtp_info);
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void clearPackets();
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void reset(double global_start_time);
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AudioRouting getAudioRouting();
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void setAudioRouting(AudioRouting audio_routing);
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#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
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void decode(QAudioDevice out_device);
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#else
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void decode(QAudioDeviceInfo out_device);
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#endif
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double startRelTime() const { return start_rel_time_; }
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double stopRelTime() const { return stop_rel_time_; }
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unsigned sampleRate() const { return first_sample_rate_; }
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unsigned playRate() const { return audio_out_rate_; }
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void setRequestedPlayRate(unsigned new_rate) { audio_requested_out_rate_ = new_rate; }
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const QStringList payloadNames() const;
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/**
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* @brief Return a list of visual timestamps.
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* @return A set of timestamps suitable for passing to QCPGraph::setData.
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*/
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const QVector<double> visualTimestamps(bool relative = true);
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/**
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* @brief Return a list of visual samples. There will be fewer visual samples
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* per second (1000) than the actual audio.
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* @param y_offset Y axis offset to be used for stacking graphs.
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* @return A set of values suitable for passing to QCPGraph::setData.
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*/
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const QVector<double> visualSamples(int y_offset = 0);
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/**
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* @brief Return a list of out-of-sequence timestamps.
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* @return A set of timestamps suitable for passing to QCPGraph::setData.
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*/
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const QVector<double> outOfSequenceTimestamps(bool relative = true);
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int outOfSequence() { return static_cast<int>(out_of_seq_timestamps_.size()); }
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/**
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* @brief Return a list of out-of-sequence samples. Y value is constant.
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* @param y_offset Y axis offset to be used for stacking graphs.
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* @return A set of values suitable for passing to QCPGraph::setData.
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*/
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const QVector<double> outOfSequenceSamples(int y_offset = 0);
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/**
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* @brief Return a list of jitter dropped timestamps.
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* @return A set of timestamps suitable for passing to QCPGraph::setData.
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*/
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const QVector<double> jitterDroppedTimestamps(bool relative = true);
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int jitterDropped() { return static_cast<int>(jitter_drop_timestamps_.size()); }
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/**
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* @brief Return a list of jitter dropped samples. Y value is constant.
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* @param y_offset Y axis offset to be used for stacking graphs.
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* @return A set of values suitable for passing to QCPGraph::setData.
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*/
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const QVector<double> jitterDroppedSamples(int y_offset = 0);
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/**
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* @brief Return a list of wrong timestamps.
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* @return A set of timestamps suitable for passing to QCPGraph::setData.
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*/
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const QVector<double> wrongTimestampTimestamps(bool relative = true);
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int wrongTimestamps() { return static_cast<int>(wrong_timestamp_timestamps_.size()); }
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/**
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* @brief Return a list of wrong timestamp samples. Y value is constant.
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* @param y_offset Y axis offset to be used for stacking graphs.
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* @return A set of values suitable for passing to QCPGraph::setData.
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*/
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const QVector<double> wrongTimestampSamples(int y_offset = 0);
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/**
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* @brief Return a list of inserted silence timestamps.
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* @return A set of timestamps suitable for passing to QCPGraph::setData.
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*/
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const QVector<double> insertedSilenceTimestamps(bool relative = true);
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int insertedSilences() { return static_cast<int>(silence_timestamps_.size()); }
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/**
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* @brief Return a list of wrong timestamp samples. Y value is constant.
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* @param y_offset Y axis offset to be used for stacking graphs.
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* @return A set of values suitable for passing to QCPGraph::setData.
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*/
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const QVector<double> insertedSilenceSamples(int y_offset = 0);
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quint32 nearestPacket(double timestamp, bool is_relative = true);
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QRgb color() { return color_; }
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void setColor(QRgb color) { color_ = color; }
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QAudio::State outputState() const;
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void setJitterBufferSize(int jitter_buffer_size) { jitter_buffer_size_ = jitter_buffer_size; }
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void setTimingMode(TimingMode timing_mode) { timing_mode_ = timing_mode; }
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void setStartPlayTime(double start_play_time) { start_play_time_ = start_play_time; }
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#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
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bool prepareForPlay(QAudioDevice out_device);
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#else
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bool prepareForPlay(QAudioDeviceInfo out_device);
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#endif
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void startPlaying();
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void pausePlaying();
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void stopPlaying();
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void seekPlaying(qint64 samples);
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void setStereoRequired(bool stereo_required) { stereo_required_ = stereo_required; }
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qint16 getMaxSampleValue() { return max_sample_val_; }
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void setMaxSampleValue(gint16 max_sample_val) { max_sample_val_used_ = max_sample_val; }
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void seekSample(qint64 samples);
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qint64 readSample(SAMPLE *sample);
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qint64 getLeadSilenceSamples() { return prepend_samples_; }
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qint64 getTotalSamples() { return (audio_file_->getTotalSamples()); }
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qint64 getEndOfSilenceSample() { return (audio_file_->getEndOfSilenceSample()); }
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double getEndOfSilenceTime() { return (double)getEndOfSilenceSample() / (double)playRate(); }
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qint64 convertTimeToSamples(double time) { return (qint64)(time * playRate()); }
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bool savePayload(QIODevice *file);
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guint getHash() { return rtpstream_id_to_hash(&(id_)); }
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rtpstream_id_t *getID() { return &(id_); }
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QString getIDAsQString();
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rtpstream_info_t *getStreamInfo() { return &rtpstream_; }
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signals:
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void processedSecs(double secs);
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void playbackError(const QString error_msg);
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void finishedPlaying(RtpAudioStream *stream, QAudio::Error error);
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private:
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// Used to identify unique streams.
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// The GTK+ UI also uses the call number + current channel.
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rtpstream_id_t id_;
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rtpstream_info_t rtpstream_;
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bool first_packet_;
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QVector<struct _rtp_packet *>rtp_packets_;
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RtpAudioFile *audio_file_; // Stores waveform samples in sparse file
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QIODevice *temp_file_;
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struct _GHashTable *decoders_hash_;
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double global_start_rel_time_;
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double start_abs_offset_;
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double start_rel_time_;
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double stop_rel_time_;
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qint64 prepend_samples_; // Count of silence samples at begin of the stream to align with other streams
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AudioRouting audio_routing_;
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bool stereo_required_;
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quint32 first_sample_rate_;
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quint32 audio_out_rate_;
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quint32 audio_requested_out_rate_;
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QSet<QString> payload_names_;
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struct SpeexResamplerState_ *audio_resampler_;
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struct SpeexResamplerState_ *visual_resampler_;
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QMap<double, quint32> packet_timestamps_;
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QVector<qint16> visual_samples_;
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QVector<double> out_of_seq_timestamps_;
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QVector<double> jitter_drop_timestamps_;
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QVector<double> wrong_timestamp_timestamps_;
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QVector<double> silence_timestamps_;
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qint16 max_sample_val_;
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qint16 max_sample_val_used_;
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QRgb color_;
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int jitter_buffer_size_;
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TimingMode timing_mode_;
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double start_play_time_;
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const QString formatDescription(const QAudioFormat & format);
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QString currentOutputDevice();
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#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
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QAudioSink *audio_output_;
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void decodeAudio(QAudioDevice out_device);
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quint32 calculateAudioOutRate(QAudioDevice out_device, unsigned int sample_rate, unsigned int requested_out_rate);
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#else
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QAudioOutput *audio_output_;
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void decodeAudio(QAudioDeviceInfo out_device);
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quint32 calculateAudioOutRate(QAudioDeviceInfo out_device, unsigned int sample_rate, unsigned int requested_out_rate);
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#endif
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void decodeVisual();
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SAMPLE *resizeBufferIfNeeded(SAMPLE *buff, gint32 *buff_bytes, qint64 requested_size);
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private slots:
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void outputStateChanged(QAudio::State new_state);
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void delayedStopStream();
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};
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#endif // QT_MULTIMEDIA_LIB
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#endif // RTPAUDIOSTREAM_H
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