Fixes for qaudiooutput on gstreamer

Add a queue element into the pipeline to decouple
things between input and output. Add ability to
suspend/resume the QGstAppSrc.

This finally makes the autotest pass consistently.

Change-Id: Ib20d1f9e6932aa9459085ad46fe78fdd03749723
Reviewed-by: Doris Verria <doris.verria@qt.io>
Reviewed-by: Lars Knoll <lars.knoll@qt.io>

4 files changed, 17 insertions, 5 deletions

diff --git a/src/multimedia/platform/gstreamer/audio/qaudiooutput_gstreamer.cpp b/src/multimedia/platform/gstreamer/audio/qaudiooutput_gstreamer.cpp
index 1b0fec46a..1d1e94ff6 100644
--- a/src/multimedia/platform/gstreamer/audio/qaudiooutput_gstreamer.cpp
+++ b/src/multimedia/platform/gstreamer/audio/qaudiooutput_gstreamer.cpp
@@ -67,6 +67,7 @@ QGStreamerAudioOutput::QGStreamerAudioOutput(const QAudioDeviceInfo &device)
     gstAppSrc = m_appSrc->element();
 
     //    gstDecodeBin = gst_element_factory_make ("decodebin", "dec");
+    QGstElement queue("queue", "queue");
     QGstElement conv("audioconvert", "conv");
     gstVolume = QGstElement("volume", "volume");
     if (m_volume != 1.)
@@ -78,8 +79,8 @@ QGStreamerAudioOutput::QGStreamerAudioOutput(const QAudioDeviceInfo &device)
     const auto *audioInfo = static_cast<const QGStreamerAudioDeviceInfo *>(device.handle());
     gstOutput = gst_device_create_element(audioInfo->gstDevice, nullptr);
 
-    gstPipeline.add(gstAppSrc, /*gstDecodeBin, */ conv, gstVolume, gstOutput);
-    gstAppSrc.link(conv, gstVolume, gstOutput);
+    gstPipeline.add(gstAppSrc, queue, /*gstDecodeBin, */ conv, gstVolume, gstOutput);
+    gstAppSrc.link(queue, conv, gstVolume, gstOutput);
 }
 
 QGStreamerAudioOutput::~QGStreamerAudioOutput()
@@ -258,6 +259,7 @@ qint64 QGStreamerAudioOutput::write(const char *data, qint64 len)
         return 0;
     if (m_errorState == QAudio::UnderrunError)
         m_errorState = QAudio::NoError;
+
     m_appSrc->write(data, len);
     return len;
 }
@@ -305,6 +307,7 @@ qint64 QGStreamerAudioOutput::processedUSecs() const
 void QGStreamerAudioOutput::resume()
 {
     if (m_deviceState == QAudio::SuspendedState) {
+        m_appSrc->resume();
         gstPipeline.setState(GST_STATE_PLAYING);
 
         setState(m_pullMode ? QAudio::ActiveState : QAudio::IdleState);
@@ -329,6 +332,7 @@ void QGStreamerAudioOutput::suspend()
         setState(QAudio::SuspendedState);
 
         gstPipeline.setState(GST_STATE_PAUSED);
+        m_appSrc->suspend();
         // ### elapsed time
     }
 }
diff --git a/src/multimedia/platform/gstreamer/common/qgstappsrc.cpp b/src/multimedia/platform/gstreamer/common/qgstappsrc.cpp
index 801db5aff..027cc1850 100644
--- a/src/multimedia/platform/gstreamer/common/qgstappsrc.cpp
+++ b/src/multimedia/platform/gstreamer/common/qgstappsrc.cpp
@@ -78,6 +78,7 @@ bool QGstAppSrc::setup(QIODevice *stream, qint64 offset)
     gst_app_src_set_callbacks(appSrc, (GstAppSrcCallbacks*)&m_callbacks, this, nullptr);
 
     m_maxBytes = gst_app_src_get_max_bytes(appSrc);
+    m_suspended = false;
 
     if (m_sequential)
         m_streamType = GST_APP_STREAM_TYPE_STREAM;
@@ -87,6 +88,7 @@ bool QGstAppSrc::setup(QIODevice *stream, qint64 offset)
     gst_app_src_set_size(appSrc, m_sequential ? -1 : m_stream->size() - m_offset);
 
     m_networkReply = qobject_cast<QNetworkReply *>(m_stream);
+    m_noMoreData = true;
 
     return true;
 }
@@ -164,8 +166,10 @@ void QGstAppSrc::streamDestroyed()
 
 void QGstAppSrc::pushData()
 {
-    if (m_appSrc.isNull() || !m_dataRequestSize)
+    if (m_appSrc.isNull() || !m_dataRequestSize || m_suspended) {
+        qCDebug(qLcAppSrc) << "push data: return immediately" << m_appSrc.isNull() << m_dataRequestSize << m_suspended;
         return;
+    }
 
     qCDebug(qLcAppSrc) << "pushData" << (m_stream ? m_stream : nullptr) << m_buffer.size();
     if ((m_stream && m_stream->atEnd())) {
diff --git a/src/multimedia/platform/gstreamer/common/qgstappsrc_p.h b/src/multimedia/platform/gstreamer/common/qgstappsrc_p.h
index 4a1ad5003..3a978454c 100644
--- a/src/multimedia/platform/gstreamer/common/qgstappsrc_p.h
+++ b/src/multimedia/platform/gstreamer/common/qgstappsrc_p.h
@@ -57,6 +57,7 @@
 #include <QtCore/qobject.h>
 #include <QtCore/qiodevice.h>
 #include <QtCore/private/qringbuffer_p.h>
+#include <QtCore/qatomic.h>
 
 #include <private/qgst_p.h>
 #include <gst/app/gstappsrc.h>
@@ -82,6 +83,9 @@ public:
 
     bool canAcceptMoreData() { return m_noMoreData || m_dataRequestSize != 0; }
 
+    void suspend() { m_suspended = true; }
+    void resume() { m_suspended = false; m_noMoreData = true; }
+
 Q_SIGNALS:
     void bytesProcessed(int bytes);
     void noMoreData();
@@ -113,12 +117,13 @@ private:
 
     QGstElement m_appSrc;
     bool m_sequential = true;
+    bool m_suspended = false;
     bool m_noMoreData = false;
     GstAppStreamType m_streamType = GST_APP_STREAM_TYPE_RANDOM_ACCESS;
     qint64 m_offset = 0;
     qint64 m_maxBytes = 0;
     qint64 bytesReadSoFar = 0;
-    unsigned int m_dataRequestSize = 0;
+    QAtomicInteger<unsigned int> m_dataRequestSize = 0;
     int streamedSamples = 0;
 };
 
diff --git a/tests/auto/integration/qaudiooutput/tst_qaudiooutput.cpp b/tests/auto/integration/qaudiooutput/tst_qaudiooutput.cpp
index 2cd59f472..8b20649b9 100644
--- a/tests/auto/integration/qaudiooutput/tst_qaudiooutput.cpp
+++ b/tests/auto/integration/qaudiooutput/tst_qaudiooutput.cpp
@@ -626,7 +626,6 @@ void tst_QAudioOutput::pushSuspendResume()
 
     qint64 written = 0;
     bool firstBuffer = true;
-    QByteArray buffer(AUDIO_BUFFER, 0);
 
     // Play half of the clip
     while (written < (audioFile->size() - QWaveDecoder::headerLength()) / 2) {