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>

5 files changed, 219 insertions, 27 deletions

diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index 443f18d1af..b729ab4f62 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -133,10 +133,13 @@ void RtpAudioStream::addRtpPacket(const struct _packet_info *pinfo, const struct
     rtp_packets_ << rtp_packet;
 }
 
-void RtpAudioStream::reset(double global_start_time)
+void RtpAudioStream::reset(double global_start_time, bool stereo, bool left, bool right)
 {
     global_start_rel_time_ = global_start_time;
     stop_rel_time_ = start_rel_time_;
+    audio_stereo_ = stereo;
+    audio_left_ = left;
+    audio_right_ = right;
     audio_out_rate_ = 0;
     max_sample_val_ = 1;
     packet_timestamps_.clear();
@@ -157,7 +160,7 @@ static const int sample_bytes_ = sizeof(SAMPLE) / sizeof(char);
 /* Fix for bug 4119/5902: don't insert too many silence frames.
  * XXX - is there a better thing to do here?
  */
-static const int max_silence_samples_ = MAX_SILENCE_FRAMES;
+static const qint64 max_silence_samples_ = MAX_SILENCE_FRAMES;
 
 void RtpAudioStream::decode()
 {
@@ -238,7 +241,11 @@ void RtpAudioStream::decode()
             format.setSampleRate(sample_rate);
             format.setSampleSize(sample_bytes_ * 8); // bits
             format.setSampleType(QAudioFormat::SignedInt);
-            format.setChannelCount(1);
+            if (audio_stereo_) {
+                format.setChannelCount(2);
+            } else {
+                format.setChannelCount(1);
+            }
             format.setCodec("audio/pcm");
 
             if (!cur_out_device.isFormatSupported(format)) {
@@ -250,7 +257,7 @@ void RtpAudioStream::decode()
 
             // Prepend silence to match our sibling streams.
             tempfile_->seek(0);
-            int prepend_samples = (start_rel_time_ - global_start_rel_time_) * audio_out_rate_;
+            qint64 prepend_samples = (start_rel_time_ - global_start_rel_time_) * audio_out_rate_;
             if (prepend_samples > 0) {
                 writeSilence(prepend_samples);
             }
@@ -278,10 +285,10 @@ void RtpAudioStream::decode()
 
             /* if there was a silence period (more than two packetization period) resync the source */
             if ((rtp_time - rtp_time_prev) > pack_period*2) {
-                int silence_samples;
+                qint64 silence_samples;
                 RTP_STREAM_DEBUG("Resync...");
 
-                silence_samples = (int)((arrive_time - arrive_time_prev)*sample_rate - decoded_bytes_prev / sample_bytes_);
+                silence_samples = (qint64)((arrive_time - arrive_time_prev)*sample_rate - decoded_bytes_prev / sample_bytes_);
                 /* Fix for bug 4119/5902: don't insert too many silence frames.
                  * XXX - is there a better thing to do here?
                  */
@@ -301,7 +308,7 @@ void RtpAudioStream::decode()
         } else {
             // rtp_player.c:664
             /* Add silence if it is necessary */
-            int silence_samples;
+            qint64 silence_samples;
 
             if (timing_mode_ == Uninterrupted) {
                 silence_samples = 0;
@@ -365,7 +372,29 @@ void RtpAudioStream::decode()
         }
 
         // Write the decoded, possibly-resampled audio to our temp file.
-        tempfile_->write(write_buff, write_bytes);
+        gint64 silence = 0;
+        if (audio_stereo_) {
+            // Process audio mute/left/right settings
+            for(qint64 i=0; i<write_bytes; i+=sample_bytes_) {
+                if (audio_left_) {
+                    tempfile_->write(write_buff+i, sample_bytes_);
+                } else {
+                    tempfile_->write((char *)&silence, sample_bytes_);
+                }
+                if (audio_right_) {
+                    tempfile_->write(write_buff+i, sample_bytes_);
+                } else {
+                    tempfile_->write((char *)&silence, sample_bytes_);
+                }
+            }
+        } else {
+            // Process audio mute/unmute settings
+            if (audio_left_) {
+                tempfile_->write(write_buff, write_bytes);
+            } else {
+                writeSilence(write_bytes / sample_bytes_);
+            }
+        }
 
         // Collect our visual samples.
         spx_uint32_t in_len = (spx_uint32_t)rtp_packet->info->info_payload_len;
@@ -570,7 +599,11 @@ void RtpAudioStream::startPlaying()
     format.setSampleRate(audio_out_rate_);
     format.setSampleSize(sample_bytes_ * 8); // bits
     format.setSampleType(QAudioFormat::SignedInt);
-    format.setChannelCount(1);
+    if (audio_stereo_) {
+        format.setChannelCount(2);
+    } else {
+        format.setChannelCount(1);
+    }
     format.setCodec("audio/pcm");
 
     // RTP_STREAM_DEBUG("playing %s %d samples @ %u Hz",
@@ -592,6 +625,10 @@ void RtpAudioStream::startPlaying()
     start_pos = (qint64)(start_play_time_ * sample_bytes_ * audio_out_rate_);
     // Round to sample_bytes_ boundary
     start_pos = (start_pos / sample_bytes_) * sample_bytes_;
+    if (audio_stereo_) {
+        // There is 2x more samples for stereo
+        start_pos *= 2;
+    }
     if (start_pos < tempfile_->size()) {
         tempfile_->seek(start_pos);
         audio_output_->start(tempfile_);
@@ -614,15 +651,21 @@ void RtpAudioStream::stopPlaying()
     }
 }
 
-void RtpAudioStream::writeSilence(int samples)
+void RtpAudioStream::writeSilence(qint64 samples)
 {
     if (samples < 1 || audio_out_rate_ == 0) return;
 
-    unsigned silence_bytes = samples * sample_bytes_;
+    qint64 silence_bytes = samples * sample_bytes_;
     char *silence_buff = (char *) g_malloc0(silence_bytes);
 
-    RTP_STREAM_DEBUG("Writing %u silence samples", samples);
-    tempfile_->write(silence_buff, silence_bytes);
+    RTP_STREAM_DEBUG("Writing %llu silence samples", samples);
+    if (audio_stereo_) {
+        // Silence for left and right channel
+        tempfile_->write(silence_buff, silence_bytes);
+        tempfile_->write(silence_buff, silence_bytes);
+    } else {
+        tempfile_->write(silence_buff, silence_bytes);
+    }
     g_free(silence_buff);
 
     // Silence is inserted to audio file only.
diff --git a/ui/qt/rtp_audio_stream.h b/ui/qt/rtp_audio_stream.h
index 64948065a1..f39df7dd72 100644
--- a/ui/qt/rtp_audio_stream.h
+++ b/ui/qt/rtp_audio_stream.h
@@ -45,7 +45,7 @@ public:
     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);
+    void reset(double global_start_time, bool stereo, bool left, bool right);
     void decode();
 
     double startRelTime() const { return start_rel_time_; }
@@ -153,6 +153,9 @@ private:
     double start_abs_offset_;
     double start_rel_time_;
     double stop_rel_time_;
+    bool audio_stereo_;
+    bool audio_left_;
+    bool audio_right_;
     quint32 audio_out_rate_;
     QSet<QString> payload_names_;
     struct SpeexResamplerState_ *audio_resampler_;
@@ -171,7 +174,7 @@ private:
     TimingMode timing_mode_;
     double start_play_time_;
 
-    void writeSilence(int samples);
+    void writeSilence(qint64 samples);
     const QString formatDescription(const QAudioFormat & format);
     QString currentOutputDevice();
 
diff --git a/ui/qt/rtp_player_dialog.cpp b/ui/qt/rtp_player_dialog.cpp
index 3be1d397df..18026df063 100644
--- a/ui/qt/rtp_player_dialog.cpp
+++ b/ui/qt/rtp_player_dialog.cpp
@@ -68,6 +68,7 @@
 // In some places we match by conv/call number, in others we match by first frame.
 
 enum {
+    channel_col_,
     src_addr_col_,
     src_port_col_,
     dst_addr_col_,
@@ -80,7 +81,8 @@ enum {
     payload_col_,
 
     stream_data_col_ = src_addr_col_, // RtpAudioStream
-    graph_data_col_ = src_port_col_ // QCPGraph
+    graph_data_col_ = src_port_col_, // QCPGraph
+    channel_data_col_ = channel_col_, // channel_mode_t
 };
 
 #ifdef QT_MULTIMEDIA_LIB
@@ -99,6 +101,7 @@ RtpPlayerDialog::RtpPlayerDialog(QWidget &parent, CaptureFile &cf) :
 #endif // QT_MULTIMEDIA_LIB
     , number_ticker_(new QCPAxisTicker)
     , datetime_ticker_(new QCPAxisTickerDateTime)
+    , stereo_available_(false)
 {
     ui->setupUi(this);
     setWindowTitle(wsApp->windowTitleString(tr("RTP Player")));
@@ -181,6 +184,8 @@ RtpPlayerDialog::RtpPlayerDialog(QWidget &parent, CaptureFile &cf) :
                 );
     ui->audioPlot->setFocus();
 
+    stereo_available_ = isStereoAvailable();
+
     QTimer::singleShot(0, this, SLOT(retapPackets()));
 #endif // QT_MULTIMEDIA_LIB
 }
@@ -239,9 +244,35 @@ void RtpPlayerDialog::rescanPackets(bool rescale_axes)
     int row_count = ui->streamTreeWidget->topLevelItemCount();
     // Clear existing graphs and reset stream values
     for (int row = 0; row < row_count; row++) {
+        bool left, right;
+
         QTreeWidgetItem *ti = ui->streamTreeWidget->topLevelItem(row);
         RtpAudioStream *audio_stream = ti->data(stream_data_col_, Qt::UserRole).value<RtpAudioStream*>();
-        audio_stream->reset(first_stream_rel_start_time_);
+        channel_mode_t channel_mode = (channel_mode_t)ti->data(channel_data_col_, Qt::UserRole).toUInt();
+        left = right = true;
+        switch (channel_mode) {
+            case channel_none:
+                left = false;
+                right = false;
+                break;
+            case channel_mono:
+                left = true;
+                right = false;
+                break;
+            case channel_stereo_left:
+                left = true;
+                right = false;
+                break;
+            case channel_stereo_right:
+                left = false;
+                right = true;
+                break;
+            case channel_stereo_both:
+                left = true;
+                right = true;
+                break;
+        }
+        audio_stream->reset(first_stream_rel_start_time_, stereo_available_, left, right);
 
         ti->setData(graph_data_col_, Qt::UserRole, QVariant());
     }
@@ -259,6 +290,7 @@ void RtpPlayerDialog::rescanPackets(bool rescale_axes)
     for (int row = 0; row < row_count; row++) {
         QTreeWidgetItem *ti = ui->streamTreeWidget->topLevelItem(row);
         RtpAudioStream *audio_stream = ti->data(stream_data_col_, Qt::UserRole).value<RtpAudioStream*>();
+        channel_mode_t channel_mode = (channel_mode_t)ti->data(channel_data_col_, Qt::UserRole).toUInt();
         int y_offset = row_count - row - 1;
 
         audio_stream->setJitterBufferSize((int) ui->jitterSpinBox->value());
@@ -282,6 +314,10 @@ void RtpPlayerDialog::rescanPackets(bool rescale_axes)
         QCPGraph *audio_graph = ui->audioPlot->addGraph();
         QPen wf_pen(audio_stream->color());
         wf_pen.setWidthF(wf_graph_normal_width_);
+        if (channel_mode == channel_none) {
+            // Indicate that audio will not be hearable
+            wf_pen.setStyle(Qt::DotLine);
+        }
         audio_graph->setPen(wf_pen);
         audio_graph->setSelectable(QCP::stNone);
         audio_graph->setData(audio_stream->visualTimestamps(relative_timestamps), audio_stream->visualSamples(y_offset));
@@ -371,6 +407,8 @@ void RtpPlayerDialog::rescanPackets(bool rescale_axes)
 
 void RtpPlayerDialog::addRtpStream(rtpstream_info_t *rtpstream)
 {
+    channel_mode_t channel_mode = channel_none;
+
     if (!rtpstream) return;
 
     // Find the RTP streams associated with this conversation.
@@ -401,6 +439,17 @@ void RtpPlayerDialog::addRtpStream(rtpstream_info_t *rtpstream)
         ti->setText(num_pkts_col_, QString::number(rtpstream->packet_count));
 
         ti->setData(stream_data_col_, Qt::UserRole, QVariant::fromValue(audio_stream));
+        if (stereo_available_) {
+            if (tli_count%2) {
+                channel_mode = channel_stereo_right;
+            } else {
+                channel_mode = channel_stereo_left;
+            }
+        } else {
+            channel_mode = channel_mono;
+        }
+        ti->setToolTip(channel_data_col_, QString(tr("Double click to change audio routing")));
+        setChannelMode(ti, channel_mode);
 
         for (int col = 0; col < ui->streamTreeWidget->columnCount(); col++) {
             QBrush fgBrush = ti->foreground(col);
@@ -750,15 +799,26 @@ void RtpPlayerDialog::on_streamTreeWidget_itemSelectionChanged()
     ui->audioPlot->setFocus();
 }
 
-const QString RtpPlayerDialog::getFormatedTime(double time)
+// Change channel if clicked channel column
+void RtpPlayerDialog::on_streamTreeWidget_itemDoubleClicked(QTreeWidgetItem *item, const int column)
+{
+    if (column == channel_col_) {
+        channel_mode_t channel_mode = (channel_mode_t)item->data(channel_data_col_, Qt::UserRole).toUInt();
+        channel_mode = changeChannelMode(channel_mode);
+        setChannelMode(item, channel_mode);
+        rescanPackets();
+    }
+}
+
+const QString RtpPlayerDialog::getFormatedTime(double f_time)
 {
     QString time_str;
 
     if (ui->todCheckBox->isChecked()) {
-        QDateTime date_time = QDateTime::fromMSecsSinceEpoch(time * 1000.0);
+        QDateTime date_time = QDateTime::fromMSecsSinceEpoch(f_time * 1000.0);
         time_str = date_time.toString("yyyy-MM-dd hh:mm:ss.zzz");
     } else {
-        time_str = QString::number(time, 'f', 3);
+        time_str = QString::number(f_time, 'f', 3);
         time_str += " s";
     }
 
@@ -859,15 +919,15 @@ void RtpPlayerDialog::drawStartPlayMarker()
     updateHintLabel();
 }
 
-void RtpPlayerDialog::setStartPlayMarker(double time)
+void RtpPlayerDialog::setStartPlayMarker(double new_time)
 {
     if (ui->todCheckBox->isChecked()) {
-        time = qBound(first_stream_abs_start_time_, time, first_stream_abs_start_time_ + streams_length_);
+        new_time = qBound(first_stream_abs_start_time_, new_time, first_stream_abs_start_time_ + streams_length_);
         // start_play_time is relative, we must calculate it
-        start_marker_time_ = time - first_stream_abs_start_time_;
+        start_marker_time_ = new_time - first_stream_abs_start_time_;
     } else {
-        time = qBound(first_stream_rel_start_time_, time, first_stream_rel_start_time_ + streams_length_);
-        start_marker_time_ = time;
+        new_time = qBound(first_stream_rel_start_time_, new_time, first_stream_rel_start_time_ + streams_length_);
+        start_marker_time_ = new_time;
     }
 }
 
@@ -892,6 +952,73 @@ void RtpPlayerDialog::updateStartStopTime(rtpstream_info_t *rtpstream, int tli_c
     streams_length_ = first_stream_rel_stop_time_ - first_stream_rel_start_time_;
 }
 
+void RtpPlayerDialog::setChannelMode(QTreeWidgetItem *ti, channel_mode_t channel_mode)
+{
+    QString t;
+
+    ti->setData(channel_data_col_, Qt::UserRole, QVariant(channel_mode));
+    switch (channel_mode) {
+        case channel_none:
+            t=QString("Mute");
+            break;
+        case channel_mono:
+            t=QString("Play");
+            break;
+        case channel_stereo_left:
+            t=QString("L");
+            break;
+        case channel_stereo_right:
+            t=QString("R");
+            break;
+        case channel_stereo_both:
+            t=QString("L+R");
+            break;
+    }
+
+    ti->setText(channel_col_, t);
+}
+
+channel_mode_t RtpPlayerDialog::changeChannelMode(channel_mode_t channel_mode)
+{
+    if (stereo_available_) {
+        // Stereo
+        switch (channel_mode) {
+            case channel_stereo_left:
+                return channel_stereo_both;
+            case channel_stereo_both:
+                return channel_stereo_right;
+            case channel_stereo_right:
+                return channel_none;
+            case channel_none:
+                return channel_stereo_left;
+            default:
+                return channel_stereo_left;
+        }
+    } else {
+        // Mono
+        switch (channel_mode) {
+            case channel_none:
+                return channel_mono;
+            case channel_mono:
+                return channel_none;
+            default:
+                return channel_mono;
+        }
+    }
+}
+
+bool RtpPlayerDialog::isStereoAvailable()
+{
+    QAudioDeviceInfo cur_out_device = QAudioDeviceInfo::defaultOutputDevice();
+    foreach(int count, cur_out_device.supportedChannelCounts()) {
+        if (count>1) {
+            return true;
+        }
+    }
+
+    return false;
+}
+
 #if 0
 // This also serves as a title in RtpAudioFrame.
 static const QString stream_key_tmpl_ = "%1:%2 " UTF8_RIGHTWARDS_ARROW " %3:%4 0x%5";
diff --git a/ui/qt/rtp_player_dialog.h b/ui/qt/rtp_player_dialog.h
index d30066e144..b7035ebeaa 100644
--- a/ui/qt/rtp_player_dialog.h
+++ b/ui/qt/rtp_player_dialog.h
@@ -19,11 +19,20 @@
 #include "wireshark_dialog.h"
 
 #include <QMap>
+#include <QTreeWidgetItem>
 
 namespace Ui {
 class RtpPlayerDialog;
 }
 
+typedef enum {
+    channel_none,         // Mute
+    channel_mono,         // Play
+    channel_stereo_left,  // L
+    channel_stereo_right, // R
+    channel_stereo_both   // L+R
+} channel_mode_t;
+
 class QCPItemStraightLine;
 class QDialogButtonBox;
 class QMenu;
@@ -104,6 +113,7 @@ private slots:
     void on_actionMoveRight1_triggered();
     void on_actionGoToPacket_triggered();
     void on_streamTreeWidget_itemSelectionChanged();
+    void on_streamTreeWidget_itemDoubleClicked(QTreeWidgetItem *item, const int column);
     void on_outputDeviceComboBox_currentIndexChanged(const QString &);
     void on_jitterSpinBox_valueChanged(double);
     void on_timingComboBox_currentIndexChanged(int);
@@ -123,6 +133,7 @@ private:
     QString playback_error_;
     QSharedPointer<QCPAxisTicker> number_ticker_;
     QSharedPointer<QCPAxisTickerDateTime> datetime_ticker_;
+    bool stereo_available_;
 
 //    const QString streamKey(const rtpstream_info_t *rtpstream);
 //    const QString streamKey(const packet_info *pinfo, const struct _rtp_info *rtpinfo);
@@ -135,14 +146,17 @@ private:
     void addPacket(packet_info *pinfo, const struct _rtp_info *rtpinfo);
     void zoomXAxis(bool in);
     void panXAxis(int x_pixels);
-    const QString getFormatedTime(double time);
+    const QString getFormatedTime(double f_time);
     const QString getFormatedHoveredTime();
     int getHoveredPacket();
     QString currentOutputDeviceName();
     double getStartPlayMarker();
     void drawStartPlayMarker();
-    void setStartPlayMarker(double time);
+    void setStartPlayMarker(double new_time);
     void updateStartStopTime(rtpstream_info_t *rtpstream, int tli_count);
+    void setChannelMode(QTreeWidgetItem *ti, channel_mode_t channel_mode);
+    channel_mode_t changeChannelMode(channel_mode_t channel_mode);
+    bool isStereoAvailable();
 
 #else // QT_MULTIMEDIA_LIB
 private:
diff --git a/ui/qt/rtp_player_dialog.ui b/ui/qt/rtp_player_dialog.ui
index a1e478ebd8..af0c7deb47 100644
--- a/ui/qt/rtp_player_dialog.ui
+++ b/ui/qt/rtp_player_dialog.ui
@@ -32,6 +32,11 @@
       </property>
       <column>
        <property name="text">
+        <string>Play</string>
+       </property>
+      </column>
+      <column>
+       <property name="text">
         <string>Source Address</string>
        </property>
       </column>