commit 526e39493ce9a7337a152d5ee10827d31db374e0
author eric.carlson@apple.com <eric.carlson@apple.com@268f45cc-cd09-0410-ab3c-d52691b4dbfc> Thu Oct 10 15:31:20 2019 +0000
committer eric.carlson@apple.com <eric.carlson@apple.com@268f45cc-cd09-0410-ab3c-d52691b4dbfc> Thu Oct 10 15:31:20 2019 +0000
tree e75bd1cffffc02bc05b1a2c0a51dc7277eb3217a
parent 6a78c28344c238a2e06ea3e0b14ef6a8b4fe6a33

[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.


git-svn-id: http://svn.webkit.org/repository/webkit/trunk@250971 268f45cc-cd09-0410-ab3c-d52691b4dbfc

Source/WebCore/ChangeLog

diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
index 3468950..ed3f98e 100644
--- a/Source/WebCore/ChangeLog
+++ b/Source/WebCore/ChangeLog
@@ -1,3 +1,15 @@
+2019-10-10  Eric Carlson  <eric.carlson@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  <youenn@apple.com>
 
         Remove unified plan runtime flag

Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp

diff --git a/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp b/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp
index 42cb9c6..d8b929f 100644
--- a/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp
+++ b/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp
@@ -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) {
@@ -98,30 +101,34 @@
         }
     }
 
+    void reconfigure()
+    {
+        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);
+
+        if (m_src)
+            gst_app_src_set_caps(GST_APP_SRC(m_src.get()), m_streamFormat->caps());
+
+        m_bipBopBuffer.resize(sampleCount);
+        m_bipBopBuffer.fill(0);
+
+        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);
+    }
+
     void settingsDidChange(OptionSet<RealtimeMediaSourceSettings::Flag> settings)
     {
-        if (settings.contains(RealtimeMediaSourceSettings::Flag::SampleRate)) {
-            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);
-
-            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);
-
-            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);
-        }
+        if (settings.contains(RealtimeMediaSourceSettings::Flag::SampleRate))
+            reconfigure();
 
         MockRealtimeAudioSource::settingsDidChange(settings);
     }