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/* rtp_audio_stream.h
 *
 * Wireshark - Network traffic analyzer
 * By Gerald Combs <gerald@wireshark.org>
 * Copyright 1998 Gerald Combs
 *
 * SPDX-License-Identifier: GPL-2.0-or-later
 */

#ifndef RTPAUDIOSTREAM_H
#define RTPAUDIOSTREAM_H

#include "config.h"

#ifdef QT_MULTIMEDIA_LIB

#include <glib.h>

#include <epan/address.h>
#include <ui/rtp_stream.h>
#include <ui/qt/utils/rtp_audio_routing.h>

#include <QAudio>
#include <QColor>
#include <QMap>
#include <QObject>
#include <QSet>
#include <QVector>
#include <QIODevice>
#include <QAudioOutput>

class QAudioFormat;
class QAudioOutput;
class QTemporaryFile;

struct _rtp_info;
struct _rtp_sample;

// Structure used for storing frame num during visual waveform decoding
typedef struct {
    qint64  len;
    guint32 frame_num;
} _rtp_packet_frame;

class RtpAudioStream : public QObject
{
    Q_OBJECT
public:
    enum TimingMode { JitterBuffer, RtpTimestamp, Uninterrupted };

    explicit RtpAudioStream(QObject *parent, rtpstream_info_t *rtpstream, bool stereo_required);
    ~RtpAudioStream();
    bool isMatch(const rtpstream_info_t *rtpstream) const;
    bool isMatch(const struct _packet_info *pinfo, const struct _rtp_info *rtp_info) const;
    //void addRtpStream(const rtpstream_info_t *rtpstream);
    void addRtpPacket(const struct _packet_info *pinfo, const struct _rtp_info *rtp_info);
    void reset(double global_start_time);
    AudioRouting getAudioRouting();
    void setAudioRouting(AudioRouting audio_routing);
    void decode(QAudioDeviceInfo out_device);

    double startRelTime() const { return start_rel_time_; }
    double stopRelTime() const { return stop_rel_time_; }
    unsigned sampleRate() const { return audio_out_rate_; }
    const QStringList payloadNames() const;

    /**
     * @brief Return a list of visual timestamps.
     * @return A set of timestamps suitable for passing to QCPGraph::setData.
     */
    const QVector<double> visualTimestamps(bool relative = true);
    /**
     * @brief Return a list of visual samples. There will be fewer visual samples
     * per second (1000) than the actual audio.
     * @param y_offset Y axis offset to be used for stacking graphs.
     * @return A set of values suitable for passing to QCPGraph::setData.
     */
    const QVector<double> visualSamples(int y_offset = 0);

    /**
     * @brief Return a list of out-of-sequence timestamps.
     * @return A set of timestamps suitable for passing to QCPGraph::setData.
     */
    const QVector<double> outOfSequenceTimestamps(bool relative = true);
    int outOfSequence() { return out_of_seq_timestamps_.size(); }
    /**
     * @brief Return a list of out-of-sequence samples. Y value is constant.
     * @param y_offset Y axis offset to be used for stacking graphs.
     * @return A set of values suitable for passing to QCPGraph::setData.
     */
    const QVector<double> outOfSequenceSamples(int y_offset = 0);

    /**
     * @brief Return a list of jitter dropped timestamps.
     * @return A set of timestamps suitable for passing to QCPGraph::setData.
     */
    const QVector<double> jitterDroppedTimestamps(bool relative = true);
    int jitterDropped() { return jitter_drop_timestamps_.size(); }
    /**
     * @brief Return a list of jitter dropped samples. Y value is constant.
     * @param y_offset Y axis offset to be used for stacking graphs.
     * @return A set of values suitable for passing to QCPGraph::setData.
     */
    const QVector<double> jitterDroppedSamples(int y_offset = 0);

    /**
     * @brief Return a list of wrong timestamps.
     * @return A set of timestamps suitable for passing to QCPGraph::setData.
     */
    const QVector<double> wrongTimestampTimestamps(bool relative = true);
    int wrongTimestamps() { return wrong_timestamp_timestamps_.size(); }
    /**
     * @brief Return a list of wrong timestamp samples. Y value is constant.
     * @param y_offset Y axis offset to be used for stacking graphs.
     * @return A set of values suitable for passing to QCPGraph::setData.
     */
    const QVector<double> wrongTimestampSamples(int y_offset = 0);

    /**
     * @brief Return a list of inserted silence timestamps.
     * @return A set of timestamps suitable for passing to QCPGraph::setData.
     */
    const QVector<double> insertedSilenceTimestamps(bool relative = true);
    int insertedSilences() { return silence_timestamps_.size(); }
    /**
     * @brief Return a list of wrong timestamp samples. Y value is constant.
     * @param y_offset Y axis offset to be used for stacking graphs.
     * @return A set of values suitable for passing to QCPGraph::setData.
     */
    const QVector<double> insertedSilenceSamples(int y_offset = 0);

    quint32 nearestPacket(double timestamp, bool is_relative = true);

    QRgb color() { return color_; }
    void setColor(QRgb color) { color_ = color; }

    QAudio::State outputState() const;

    void setJitterBufferSize(int jitter_buffer_size) { jitter_buffer_size_ = jitter_buffer_size; }
    void setTimingMode(TimingMode timing_mode) { timing_mode_ = timing_mode; }
    void setStartPlayTime(double start_play_time) { start_play_time_ = start_play_time; }
    bool prepareForPlay(QAudioDeviceInfo out_device);
    void startPlaying();
    void pausePlaying();
    void stopPlaying();
    void setStereoRequired(bool stereo_required) { stereo_required_ = stereo_required; }

signals:
    void processedSecs(double secs);
    void playbackError(const QString error_msg);
    void finishedPlaying(RtpAudioStream *stream);

private:
    // Used to identify unique streams.
    // The GTK+ UI also uses the call number + current channel.
    rtpstream_id_t id_;

    QVector<struct _rtp_packet *>rtp_packets_;
    QTemporaryFile *sample_file_;       // Stores waveform samples
    QTemporaryFile *sample_file_frame_; // Stores _rtp_packet_frame per packet
    QIODevice *temp_file_;
    struct _GHashTable *decoders_hash_;
    // TODO: It is not used
    //QList<const rtpstream_info_t *>rtpstreams_;
    double global_start_rel_time_;
    double start_abs_offset_;
    double start_rel_time_;
    double stop_rel_time_;
    qint64 prepend_samples_; // Count of silence samples to match other streams
    AudioRouting audio_routing_;
    bool stereo_required_;
    quint32 audio_out_rate_;
    QSet<QString> payload_names_;
    struct SpeexResamplerState_ *audio_resampler_;
    struct SpeexResamplerState_ *visual_resampler_;
    QAudioOutput *audio_output_;
    QMap<double, quint32> packet_timestamps_;
    QVector<qint16> visual_samples_;
    QVector<double> out_of_seq_timestamps_;
    QVector<double> jitter_drop_timestamps_;
    QVector<double> wrong_timestamp_timestamps_;
    QVector<double> silence_timestamps_;
    qint16 max_sample_val_;
    QRgb color_;

    int jitter_buffer_size_;
    TimingMode timing_mode_;
    double start_play_time_;

    void writeSilence(qint64 samples);
    const QString formatDescription(const QAudioFormat & format);
    QString currentOutputDevice();

    void decodeAudio(QAudioDeviceInfo out_device);
    void decodeVisual();

private slots:
    void outputStateChanged(QAudio::State new_state);
};

#endif // QT_MULTIMEDIA_LIB

#endif // RTPAUDIOSTREAM_H