Modified: trunk/Source/WebCore/ChangeLog (250970 => 250971)
--- trunk/Source/WebCore/ChangeLog 2019-10-10 15:19:46 UTC (rev 250970)
+++ trunk/Source/WebCore/ChangeLog 2019-10-10 15:31:20 UTC (rev 250971)
@@ -1,3 +1,15 @@
+2019-10-10 Eric Carlson <eric.carl...@apple.com>
+
+ [GTK][WPE] Lots of media related tests crashing or flaky after r250918 - [ Mac WK2 ] Layout Test fast/mediastream/MediaStreamTrack-getSettings.html is a flaky failure
+ https://bugs.webkit.org/show_bug.cgi?id=202784
+
+ Unreviewed, fix GTK and WPE test crashes.
+
+ * platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp:
+ (WebCore::WrappedMockRealtimeAudioSource::render): Reconfigure if necessary.
+ (WebCore::WrappedMockRealtimeAudioSource::reconfigure): New, setup for rendering.
+ (WebCore::WrappedMockRealtimeAudioSource::settingsDidChange): Call reconfigure.
+
2019-10-10 youenn fablet <you...@apple.com>
Remove unified plan runtime flag
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp (250970 => 250971)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp 2019-10-10 15:19:46 UTC (rev 250970)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp 2019-10-10 15:31:20 UTC (rev 250971)
@@ -73,6 +73,9 @@
{
ASSERT(m_src);
+ if (!m_bipBopBuffer.size())
+ reconfigure();
+
uint32_t totalFrameCount = GST_ROUND_UP_16(static_cast<size_t>(delta.seconds() * sampleRate()));
uint32_t frameCount = std::min(totalFrameCount, m_maximiumFrameCount);
while (frameCount) {
@@ -98,31 +101,35 @@
}
}
- void settingsDidChange(OptionSet<RealtimeMediaSourceSettings::Flag> settings)
+ void reconfigure()
{
- if (settings.contains(RealtimeMediaSourceSettings::Flag::SampleRate)) {
- GstAudioInfo info;
- auto rate = sampleRate();
- size_t sampleCount = 2 * rate;
+ GstAudioInfo info;
+ auto rate = sampleRate();
+ size_t sampleCount = 2 * rate;
- m_maximiumFrameCount = WTF::roundUpToPowerOfTwo(renderInterval().seconds() * sampleRate());
- gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_F32LE, rate, 1, nullptr);
- m_streamFormat = GStreamerAudioStreamDescription(info);
+ m_maximiumFrameCount = WTF::roundUpToPowerOfTwo(renderInterval().seconds() * sampleRate());
+ gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_F32LE, rate, 1, nullptr);
+ m_streamFormat = GStreamerAudioStreamDescription(info);
- if (m_src)
- gst_app_src_set_caps(GST_APP_SRC(m_src.get()), m_streamFormat->caps());
+ if (m_src)
+ gst_app_src_set_caps(GST_APP_SRC(m_src.get()), m_streamFormat->caps());
- m_bipBopBuffer.grow(sampleCount);
- m_bipBopBuffer.fill(0);
+ m_bipBopBuffer.resize(sampleCount);
+ m_bipBopBuffer.fill(0);
- size_t bipBopSampleCount = ceil(s_BipBopDuration * rate);
- size_t bipStart = 0;
- size_t bopStart = rate;
+ size_t bipBopSampleCount = ceil(s_BipBopDuration * rate);
+ size_t bipStart = 0;
+ size_t bopStart = rate;
- addHum(s_BipBopVolume, s_BipFrequency, rate, 0, static_cast<float*>(m_bipBopBuffer.data() + bipStart), bipBopSampleCount);
- addHum(s_BipBopVolume, s_BopFrequency, rate, 0, static_cast<float*>(m_bipBopBuffer.data() + bopStart), bipBopSampleCount);
- }
+ addHum(s_BipBopVolume, s_BipFrequency, rate, 0, static_cast<float*>(m_bipBopBuffer.data() + bipStart), bipBopSampleCount);
+ addHum(s_BipBopVolume, s_BopFrequency, rate, 0, static_cast<float*>(m_bipBopBuffer.data() + bopStart), bipBopSampleCount);
+ }
+ void settingsDidChange(OptionSet<RealtimeMediaSourceSettings::Flag> settings)
+ {
+ if (settings.contains(RealtimeMediaSourceSettings::Flag::SampleRate))
+ reconfigure();
+
MockRealtimeAudioSource::settingsDidChange(settings);
}
