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>

1 files changed, 25 insertions, 14 deletions

diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index fc04257bc0..443f18d1af 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -133,9 +133,9 @@ void RtpAudioStream::addRtpPacket(const struct _packet_info *pinfo, const struct
     rtp_packets_ << rtp_packet;
 }
 
-void RtpAudioStream::reset(double start_rel_time)
+void RtpAudioStream::reset(double global_start_time)
 {
-    global_start_rel_time_ = start_rel_time;
+    global_start_rel_time_ = global_start_time;
     stop_rel_time_ = start_rel_time_;
     audio_out_rate_ = 0;
     max_sample_val_ = 1;
@@ -158,6 +158,7 @@ static const int sample_bytes_ = sizeof(SAMPLE) / sizeof(char);
  * XXX - is there a better thing to do here?
  */
 static const int max_silence_samples_ = MAX_SILENCE_FRAMES;
+
 void RtpAudioStream::decode()
 {
     if (rtp_packets_.size() < 1) return;
@@ -402,7 +403,7 @@ const QVector<double> RtpAudioStream::visualTimestamps(bool relative)
 
     QVector<double> adj_timestamps;
     for (int i = 0; i < ts_keys.size(); i++) {
-        adj_timestamps.append(ts_keys[i] + start_abs_offset_);
+        adj_timestamps.append(ts_keys[i] + start_abs_offset_ - start_rel_time_);
     }
     return adj_timestamps;
 }
@@ -429,7 +430,7 @@ const QVector<double> RtpAudioStream::outOfSequenceTimestamps(bool relative)
 
     QVector<double> adj_timestamps;
     for (int i = 0; i < out_of_seq_timestamps_.size(); i++) {
-        adj_timestamps.append(out_of_seq_timestamps_[i] + start_abs_offset_);
+        adj_timestamps.append(out_of_seq_timestamps_[i] + start_abs_offset_ - start_rel_time_);
     }
     return adj_timestamps;
 }
@@ -450,7 +451,7 @@ const QVector<double> RtpAudioStream::jitterDroppedTimestamps(bool relative)
 
     QVector<double> adj_timestamps;
     for (int i = 0; i < jitter_drop_timestamps_.size(); i++) {
-        adj_timestamps.append(jitter_drop_timestamps_[i] + start_abs_offset_);
+        adj_timestamps.append(jitter_drop_timestamps_[i] + start_abs_offset_ - start_rel_time_);
     }
     return adj_timestamps;
 }
@@ -471,7 +472,7 @@ const QVector<double> RtpAudioStream::wrongTimestampTimestamps(bool relative)
 
     QVector<double> adj_timestamps;
     for (int i = 0; i < wrong_timestamp_timestamps_.size(); i++) {
-        adj_timestamps.append(wrong_timestamp_timestamps_[i] + start_abs_offset_);
+        adj_timestamps.append(wrong_timestamp_timestamps_[i] + start_abs_offset_ - start_rel_time_);
     }
     return adj_timestamps;
 }
@@ -492,7 +493,7 @@ const QVector<double> RtpAudioStream::insertedSilenceTimestamps(bool relative)
 
     QVector<double> adj_timestamps;
     for (int i = 0; i < silence_timestamps_.size(); i++) {
-        adj_timestamps.append(silence_timestamps_[i] + start_abs_offset_);
+        adj_timestamps.append(silence_timestamps_[i] + start_abs_offset_ - start_rel_time_);
     }
     return adj_timestamps;
 }
@@ -548,6 +549,8 @@ const QString RtpAudioStream::formatDescription(const QAudioFormat &format)
 
 void RtpAudioStream::startPlaying()
 {
+    qint64 start_pos;
+
     if (audio_output_) return;
 
     if (audio_out_rate_ == 0) {
@@ -586,12 +589,20 @@ void RtpAudioStream::startPlaying()
     audio_output_->setNotifyInterval(65); // ~15 fps
     connect(audio_output_, SIGNAL(stateChanged(QAudio::State)), this, SLOT(outputStateChanged(QAudio::State)));
     connect(audio_output_, SIGNAL(notify()), this, SLOT(outputNotify()));
-    tempfile_->seek(0);
-    audio_output_->start(tempfile_);
-    emit startedPlaying();
-    // QTBUG-6548 StoppedState is not always emitted on error, force a cleanup
-    // in case playback fails immediately.
-    if (audio_output_ && audio_output_->state() == QAudio::StoppedState) {
+    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 (start_pos < tempfile_->size()) {
+        tempfile_->seek(start_pos);
+        audio_output_->start(tempfile_);
+        emit startedPlaying();
+        // QTBUG-6548 StoppedState is not always emitted on error, force a cleanup
+        // in case playback fails immediately.
+        if (audio_output_ && audio_output_->state() == QAudio::StoppedState) {
+            outputStateChanged(QAudio::StoppedState);
+        }
+    } else {
+        // Report stopped audio if start position is later than stream ends
         outputStateChanged(QAudio::StoppedState);
     }
 }
@@ -648,7 +659,7 @@ void RtpAudioStream::outputStateChanged(QAudio::State new_state)
 
 void RtpAudioStream::outputNotify()
 {
-    emit processedSecs(audio_output_->processedUSecs() / 1000000.0);
+    emit processedSecs((audio_output_->processedUSecs() / 1000000.0) + start_play_time_);
 }
 
 #endif // QT_MULTIMEDIA_LIB