Qt: Update the RTP stream UI to support Qt6Multimedia.

Fixes #18115

1 files changed, 75 insertions, 5 deletions

diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index 83b3c248ea..a1cb42c2f9 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -31,6 +31,10 @@
 #include <ui/qt/utils/rtp_audio_routing_filter.h>
 #include <ui/qt/utils/rtp_audio_file.h>
 
+#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
+#include <QAudioDevice>
+#include <QAudioSink>
+#endif
 #include <QAudioFormat>
 #include <QAudioOutput>
 #include <QVariant>
@@ -54,13 +58,13 @@ RtpAudioStream::RtpAudioStream(QObject *parent, rtpstream_id_t *id, bool stereo_
     , audio_out_rate_(0)
     , audio_requested_out_rate_(0)
     , audio_resampler_(0)
-    , audio_output_(NULL)
     , max_sample_val_(1)
     , max_sample_val_used_(1)
     , color_(0)
     , jitter_buffer_size_(50)
     , timing_mode_(RtpAudioStream::JitterBuffer)
     , start_play_time_(0)
+    , audio_output_(NULL)
 {
     rtpstream_id_copy(id, &id_);
     memset(&rtpstream_, 0, sizeof(rtpstream_));
@@ -181,7 +185,11 @@ void RtpAudioStream::setAudioRouting(AudioRouting audio_routing)
     audio_routing_ = audio_routing;
 }
 
+#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
+void RtpAudioStream::decode(QAudioDevice out_device)
+#else
 void RtpAudioStream::decode(QAudioDeviceInfo out_device)
+#endif
 {
     if (rtp_packets_.size() < 1) return;
 
@@ -200,7 +208,11 @@ void RtpAudioStream::decode(QAudioDeviceInfo out_device)
 }
 
 // Side effect: it creates and initiates resampler if needed
+#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
+quint32 RtpAudioStream::calculateAudioOutRate(QAudioDevice out_device, unsigned int sample_rate, unsigned int requested_out_rate)
+#else
 quint32 RtpAudioStream::calculateAudioOutRate(QAudioDeviceInfo out_device, unsigned int sample_rate, unsigned int requested_out_rate)
+#endif
 {
     quint32 out_rate;
 
@@ -208,19 +220,30 @@ quint32 RtpAudioStream::calculateAudioOutRate(QAudioDeviceInfo out_device, unsig
     // our audio hardware.
     QAudioFormat format;
     format.setSampleRate(sample_rate);
+#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
+    // Must match rtp_media.h.
+    format.setSampleFormat(QAudioFormat::Int16);
+#else
     format.setSampleSize(SAMPLE_BYTES * 8); // bits
     format.setSampleType(QAudioFormat::SignedInt);
+#endif
     if (stereo_required_) {
         format.setChannelCount(2);
     } else {
         format.setChannelCount(1);
     }
+#if (QT_VERSION < QT_VERSION_CHECK(6, 0, 0))
     format.setCodec("audio/pcm");
+#endif
 
     if (!out_device.isFormatSupported(format) &&
         (requested_out_rate==0)
        ) {
+#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
+        out_rate = out_device.preferredFormat().sampleRate();
+#else
         out_rate = out_device.nearestFormat(format).sampleRate();
+#endif
         audio_resampler_ = speex_resampler_init(1, sample_rate, out_rate, 10, NULL);
         RTP_STREAM_DEBUG("Started resampling from %u to (out) %u Hz.", sample_rate, out_rate);
     } else {
@@ -240,7 +263,11 @@ quint32 RtpAudioStream::calculateAudioOutRate(QAudioDeviceInfo out_device, unsig
     return out_rate;
 }
 
+#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
+void RtpAudioStream::decodeAudio(QAudioDevice out_device)
+#else
 void RtpAudioStream::decodeAudio(QAudioDeviceInfo out_device)
+#endif
 {
     // XXX This is more messy than it should be.
 
@@ -607,10 +634,10 @@ const QVector<double> RtpAudioStream::insertedSilenceSamples(int y_offset)
 
 quint32 RtpAudioStream::nearestPacket(double timestamp, bool is_relative)
 {
-    if (packet_timestamps_.keys().count() < 1) return 0;
+    if (packet_timestamps_.size() < 1) return 0;
 
     if (!is_relative) timestamp -= start_abs_offset_;
-    QMap<double, quint32>::const_iterator it = packet_timestamps_.lowerBound(timestamp);
+    QMap<double, quint32>::iterator it = packet_timestamps_.lowerBound(timestamp);
     if (it == packet_timestamps_.end()) return 0;
     return it.value();
 }
@@ -624,12 +651,35 @@ QAudio::State RtpAudioStream::outputState() const
 const QString RtpAudioStream::formatDescription(const QAudioFormat &format)
 {
     QString fmt_descr = QString("%1 Hz, ").arg(format.sampleRate());
+#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
+    switch (format.sampleFormat()) {
+    case QAudioFormat::UInt8:
+        fmt_descr += "UInt8";
+        break;
+    case QAudioFormat::Int16:
+        fmt_descr += "Int16";
+        break;
+    case QAudioFormat::Int32:
+        fmt_descr += "Int32";
+        break;
+    case QAudioFormat::Float:
+        fmt_descr += "Float";
+        break;
+    default:
+        fmt_descr += "Unknown";
+        break;
+    }
+#else
     switch (format.sampleType()) {
     case QAudioFormat::SignedInt:
         fmt_descr += "Int";
+        fmt_descr += QString::number(format.sampleSize());
+        fmt_descr += format.byteOrder() == QAudioFormat::BigEndian ? "BE" : "LE";
         break;
     case QAudioFormat::UnSignedInt:
         fmt_descr += "UInt";
+        fmt_descr += QString::number(format.sampleSize());
+        fmt_descr += format.byteOrder() == QAudioFormat::BigEndian ? "BE" : "LE";
         break;
     case QAudioFormat::Float:
         fmt_descr += "Float";
@@ -638,8 +688,7 @@ const QString RtpAudioStream::formatDescription(const QAudioFormat &format)
         fmt_descr += "Unknown";
         break;
     }
-    fmt_descr += QString::number(format.sampleSize());
-    fmt_descr += format.byteOrder() == QAudioFormat::BigEndian ? "BE" : "LE";
+#endif
 
     return fmt_descr;
 }
@@ -660,7 +709,11 @@ QString RtpAudioStream::getIDAsQString()
     return str;
 }
 
+#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
+bool RtpAudioStream::prepareForPlay(QAudioDevice out_device)
+#else
 bool RtpAudioStream::prepareForPlay(QAudioDeviceInfo out_device)
+#endif
 {
     qint64 start_pos;
     qint64 size;
@@ -683,24 +736,36 @@ bool RtpAudioStream::prepareForPlay(QAudioDeviceInfo out_device)
 
     QAudioFormat format;
     format.setSampleRate(audio_out_rate_);
+#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
+    // Must match rtp_media.h.
+    format.setSampleFormat(QAudioFormat::Int16);
+#else
     format.setSampleSize(SAMPLE_BYTES * 8); // bits
     format.setSampleType(QAudioFormat::SignedInt);
+#endif
     if (stereo_required_) {
         format.setChannelCount(2);
     } else {
         format.setChannelCount(1);
     }
+#if (QT_VERSION < QT_VERSION_CHECK(6, 0, 0))
     format.setCodec("audio/pcm");
+#endif
 
     // RTP_STREAM_DEBUG("playing %s %d samples @ %u Hz",
     //                 sample_file_->fileName().toUtf8().constData(),
     //                 (int) sample_file_->size(), audio_out_rate_);
 
     if (!out_device.isFormatSupported(format)) {
+#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
+        QString playback_error = tr("%1 does not support PCM at %2. Preferred format is %3")
+                .arg(out_device.description(), formatDescription(format), formatDescription(out_device.preferredFormat()));
+#else
         QString playback_error = tr("%1 does not support PCM at %2. Preferred format is %3")
                 .arg(out_device.deviceName())
                 .arg(formatDescription(format))
                 .arg(formatDescription(out_device.nearestFormat(format)));
+#endif
         emit playbackError(playback_error);
     }
 
@@ -718,8 +783,13 @@ bool RtpAudioStream::prepareForPlay(QAudioDeviceInfo out_device)
         temp_file_ = new AudioRoutingFilter(audio_file_, stereo_required_, audio_routing_);
         temp_file_->seek(start_pos);
         if (audio_output_) delete audio_output_;
+#if (QT_VERSION >= QT_VERSION_CHECK(6, 0, 0))
+        audio_output_ = new QAudioSink(out_device, format, this);
+        connect(audio_output_, &QAudioSink::stateChanged, this, &RtpAudioStream::outputStateChanged);
+#else
         audio_output_ = new QAudioOutput(out_device, format, this);
         connect(audio_output_, &QAudioOutput::stateChanged, this, &RtpAudioStream::outputStateChanged);
+#endif
         return true;
     } else {
         // Report stopped audio if start position is later than stream ends