RTP Player dialog improved

Functional changes:
  Audio routing information is now stored in audio stream and not in table. It
is handled by separate utils/rtp_audio_routing.cpp class which is able to
convert it between mono/stereo etc.
  There is new utils/rtp_audio_routing_filter.cpp class which is able to
route mono audio stream to any audio channel.
  Sample file separated from audio stream file - sample file is generated only
during recap. So when we need new waveform, we just use existing sample file
and no recap required - it is much faster.
  Audio stream exports just mono audio. Mono audio is then expanded to stereo
and correct channel by AudioRoutingFilter during play as required. So when
audio routing is changed, no recap and no audio export is required.
  When audio stream is muted, no audio is produced nor played.
  Most of signals between RtpPlayerDialog and RtpAudioStream were removed.
Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams).
Play possition is not received from every playing stream but from independent
silence stream.
  Added Mute/Unmute function.

Optimalization:
  When audio routing is changed, just graphs are updated. No retap nor audio
decoding is required.
  When TOD is changed, just graphs are updated. No retap nor audio decoding is
required.

1 files changed, 73 insertions, 92 deletions

diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index 1a41052844..e0a0ac3312 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -26,6 +26,8 @@
 
 #include <wsutil/nstime.h>
 
+#include <ui/qt/utils/rtp_audio_routing_filter.h>
+
 #include <QAudioFormat>
 #include <QAudioOutput>
 #include <QDir>
@@ -38,16 +40,14 @@
 static spx_int16_t default_audio_sample_rate_ = 8000;
 static const spx_int16_t visual_sample_rate_ = 1000;
 
-RtpAudioStream::RtpAudioStream(QObject *parent, rtpstream_info_t *rtpstream) :
+RtpAudioStream::RtpAudioStream(QObject *parent, rtpstream_info_t *rtpstream, bool stereo_required) :
     QObject(parent),
     decoders_hash_(rtp_decoder_hash_table_new()),
     global_start_rel_time_(0.0),
     start_abs_offset_(0.0),
     start_rel_time_(0.0),
     stop_rel_time_(0.0),
-    audio_stereo_(false),
-    audio_left_(false),
-    audio_right_(false),
+    stereo_required_(stereo_required),
     audio_out_rate_(0),
     audio_resampler_(0),
     audio_output_(0),
@@ -65,8 +65,8 @@ RtpAudioStream::RtpAudioStream(QObject *parent, rtpstream_info_t *rtpstream) :
     speex_resampler_skip_zeros(visual_resampler_);
 
     QString tempname = QString("%1/wireshark_rtp_stream").arg(QDir::tempPath());
-    tempfile_ = new QTemporaryFile(tempname, this);
-    tempfile_->open();
+    samplefile_ = new QTemporaryFile(tempname, this);
+    samplefile_->open();
 
     // RTP_STREAM_DEBUG("Writing to %s", tempname.toUtf8().constData());
 }
@@ -83,6 +83,9 @@ RtpAudioStream::~RtpAudioStream()
     if (audio_resampler_) speex_resampler_destroy (audio_resampler_);
     speex_resampler_destroy (visual_resampler_);
     rtpstream_id_free(&id_);
+    if (samplefile_) delete samplefile_;
+    // tempfile_ is released by audio_output_
+    if (audio_output_) delete audio_output_;
 }
 
 bool RtpAudioStream::isMatch(const rtpstream_info_t *rtpstream) const
@@ -137,13 +140,10 @@ void RtpAudioStream::addRtpPacket(const struct _packet_info *pinfo, const struct
     rtp_packets_ << rtp_packet;
 }
 
-void RtpAudioStream::reset(double global_start_time, bool stereo, bool left, bool right)
+void RtpAudioStream::reset(double global_start_time)
 {
     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 +157,17 @@ void RtpAudioStream::reset(double global_start_time, bool stereo, bool left, boo
     if (visual_resampler_) {
         speex_resampler_reset_mem(visual_resampler_);
     }
-    tempfile_->seek(0);
+    samplefile_->seek(0);
+}
+
+AudioRouting RtpAudioStream::getAudioRouting()
+{
+    return audio_routing_;
+}
+
+void RtpAudioStream::setAudioRouting(AudioRouting audio_routing)
+{
+    audio_routing_ = audio_routing;
 }
 
 /* Fix for bug 4119/5902: don't insert too many silence frames.
@@ -165,7 +175,7 @@ void RtpAudioStream::reset(double global_start_time, bool stereo, bool left, boo
  */
 static const qint64 max_silence_samples_ = MAX_SILENCE_FRAMES;
 
-void RtpAudioStream::decode()
+void RtpAudioStream::decode(QAudioDeviceInfo out_device)
 {
     if (rtp_packets_.size() < 1) return;
 
@@ -232,34 +242,26 @@ void RtpAudioStream::decode()
 
         if (audio_out_rate_ == 0) {
             // Use the first non-zero rate we find. Ajust it to match our audio hardware.
-            QAudioDeviceInfo cur_out_device = QAudioDeviceInfo::defaultOutputDevice();
-            QString cur_out_name = parent()->property("currentOutputDeviceName").toString();
-            foreach (QAudioDeviceInfo out_device, QAudioDeviceInfo::availableDevices(QAudio::AudioOutput)) {
-                if (cur_out_name == out_device.deviceName()) {
-                    cur_out_device = out_device;
-                }
-            }
-
             QAudioFormat format;
             format.setSampleRate(sample_rate);
             format.setSampleSize(SAMPLE_BYTES * 8); // bits
             format.setSampleType(QAudioFormat::SignedInt);
-            if (audio_stereo_) {
+            if (stereo_required_) {
                 format.setChannelCount(2);
             } else {
                 format.setChannelCount(1);
             }
             format.setCodec("audio/pcm");
 
-            if (!cur_out_device.isFormatSupported(format)) {
-                sample_rate = cur_out_device.nearestFormat(format).sampleRate();
+            if (!out_device.isFormatSupported(format)) {
+                sample_rate = out_device.nearestFormat(format).sampleRate();
             }
 
             audio_out_rate_ = sample_rate;
             RTP_STREAM_DEBUG("Audio sample rate is %u", audio_out_rate_);
 
             // Prepend silence to match our sibling streams.
-            tempfile_->seek(0);
+            samplefile_->seek(0);
             qint64 prepend_samples = (start_rel_time_ - global_start_rel_time_) * audio_out_rate_;
             if (prepend_samples > 0) {
                 writeSilence(prepend_samples);
@@ -375,29 +377,7 @@ void RtpAudioStream::decode()
         }
 
         // Write the decoded, possibly-resampled audio to our temp file.
-        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);
-            }
-        }
+        samplefile_->write(write_buff, write_bytes);
 
         // Collect our visual samples.
         spx_uint32_t in_len = (spx_uint32_t)rtp_packet->info->info_payload_len;
@@ -579,30 +559,27 @@ const QString RtpAudioStream::formatDescription(const QAudioFormat &format)
     return fmt_descr;
 }
 
-void RtpAudioStream::startPlaying()
+bool RtpAudioStream::prepareForPlay(QAudioDeviceInfo out_device)
 {
     qint64 start_pos;
+    qint64 size;
+
+    if (audio_routing_.isMuted())
+        return false;
 
-    if (audio_output_) return;
+    if (audio_output_)
+        return false;
 
     if (audio_out_rate_ == 0) {
         emit playbackError(tr("RTP stream is empty or codec is unsupported."));
-        return;
-    }
-
-    QAudioDeviceInfo cur_out_device = QAudioDeviceInfo::defaultOutputDevice();
-    QString cur_out_name = parent()->property("currentOutputDeviceName").toString();
-    foreach (QAudioDeviceInfo out_device, QAudioDeviceInfo::availableDevices(QAudio::AudioOutput)) {
-        if (cur_out_name == out_device.deviceName()) {
-            cur_out_device = out_device;
-        }
+        return false;
     }
 
     QAudioFormat format;
     format.setSampleRate(audio_out_rate_);
     format.setSampleSize(SAMPLE_BYTES * 8); // bits
     format.setSampleType(QAudioFormat::SignedInt);
-    if (audio_stereo_) {
+    if (stereo_required_) {
         format.setChannelCount(2);
     } else {
         format.setChannelCount(1);
@@ -610,55 +587,67 @@ void RtpAudioStream::startPlaying()
     format.setCodec("audio/pcm");
 
     // RTP_STREAM_DEBUG("playing %s %d samples @ %u Hz",
-    //                 tempfile_->fileName().toUtf8().constData(),
-    //                 (int) tempfile_->size(), audio_out_rate_);
+    //                 samplefile_->fileName().toUtf8().constData(),
+    //                 (int) samplefile_->size(), audio_out_rate_);
 
-    if (!cur_out_device.isFormatSupported(format)) {
+    if (!out_device.isFormatSupported(format)) {
         QString playback_error = tr("%1 does not support PCM at %2. Preferred format is %3")
-                .arg(cur_out_device.deviceName())
+                .arg(out_device.deviceName())
                 .arg(formatDescription(format))
-                .arg(formatDescription(cur_out_device.nearestFormat(format)));
+                .arg(formatDescription(out_device.nearestFormat(format)));
         emit playbackError(playback_error);
     }
 
     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_) {
+    size = samplefile_->size();
+    if (stereo_required_) {
         // There is 2x more samples for stereo
         start_pos *= 2;
+        size *= 2;
     }
-    if (start_pos < tempfile_->size()) {
-        int bufferSize;
+    if (start_pos < size) {
+        int buffer_size;
 
         // Start and stop audio with no connection to UI and store buffer size
+        tempfile_ = new AudioRoutingFilter(samplefile_, stereo_required_, audio_routing_);
         tempfile_->seek(start_pos);
-        audio_output_ = new QAudioOutput(cur_out_device, format, this);
+        audio_output_ = new QAudioOutput(out_device, format, this);
         audio_output_->start(tempfile_);
-        bufferSize = audio_output_->bufferSize();
+        buffer_size = audio_output_->bufferSize();
         audio_output_->stop();
 
-        // Start audio again with doubled buffer size, UI is updated
+        // Start audio again with trippled buffer size
         tempfile_->seek(start_pos);
-        audio_output_->setBufferSize(bufferSize*2);
+        audio_output_->setBufferSize(buffer_size*3);
         audio_output_->setNotifyInterval(100); // ~15 fps
         connect(audio_output_, SIGNAL(stateChanged(QAudio::State)), this, SLOT(outputStateChanged(QAudio::State)));
-        connect(audio_output_, SIGNAL(notify()), this, SLOT(outputNotify()));
-        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);
-        }
+        return true;
     } else {
         // Report stopped audio if start position is later than stream ends
         outputStateChanged(QAudio::StoppedState);
+        return false;
     }
+
+    return false;
+}
+
+void RtpAudioStream::startPlaying()
+{
+   audio_output_->start(tempfile_);
+   // 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);
+   }
 }
 
 void RtpAudioStream::pausePlaying()
 {
+    if (audio_routing_.isMuted())
+        return;
+
     if (audio_output_) {
         if (QAudio::ActiveState == audio_output_->state()) {
             audio_output_->suspend();
@@ -670,6 +659,9 @@ void RtpAudioStream::pausePlaying()
 
 void RtpAudioStream::stopPlaying()
 {
+    if (audio_routing_.isMuted())
+        return;
+
     if (audio_output_) {
         audio_output_->stop();
     }
@@ -683,13 +675,7 @@ void RtpAudioStream::writeSilence(qint64 samples)
     char *silence_buff = (char *) g_malloc0(silence_bytes);
 
     RTP_STREAM_DEBUG("Writing " G_GUINT64_FORMAT " 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);
-    }
+    samplefile_->write(silence_buff, silence_bytes);
     g_free(silence_buff);
 
     // Silence is inserted to audio file only.
@@ -714,7 +700,7 @@ void RtpAudioStream::outputStateChanged(QAudio::State new_state)
         audio_output_->disconnect();
         audio_output_->deleteLater();
         audio_output_ = NULL;
-        emit finishedPlaying();
+        emit finishedPlaying(this);
         break;
     case QAudio::IdleState:
         audio_output_->stop();
@@ -724,11 +710,6 @@ void RtpAudioStream::outputStateChanged(QAudio::State new_state)
     }
 }
 
-void RtpAudioStream::outputNotify()
-{
-    emit processedSecs((audio_output_->processedUSecs() / 1000000.0) + start_play_time_);
-}
-
 #endif // QT_MULTIMEDIA_LIB
 
 /*