Changeset 269827 in webkit

Timestamp:

Nov 15, 2020, 7:38:57 AM (6 years ago)

Author:

Philippe Normand

Message:

[GStreamer] WebAudio to MediaStream support
​https://bugs.webkit.org/show_bug.cgi?id=218335

Reviewed by Xabier Rodriguez-Calvar.

Source/WebCore:

Provide audio samples coming from the WebAudio bus to the MediaStreamAudioSource node, using
GStreamer. This patch also moves a couple files from platform/mediastream to platform/audio,
because this is where the virtual parent classes are defined.

• Modules/webaudio/MediaStreamAudioSource.cpp:
• Modules/webaudio/MediaStreamAudioSource.h:
• Modules/webaudio/MediaStreamAudioSourceGStreamer.cpp: Added.

(WebCore::copyBusData):
(WebCore::MediaStreamAudioSource::consumeAudio):

• platform/GStreamer.cmake:
• platform/audio/gstreamer/GStreamerAudioData.h: Renamed from Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioData.h.

(isType):

• platform/audio/gstreamer/GStreamerAudioStreamDescription.h: Renamed from Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h.
• platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:

(WebCore::RealtimeOutgoingAudioSourceLibWebRTC::audioSamplesAvailable):
(WebCore::RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData):

LayoutTests:

• platform/gtk/TestExpectations: webrtc/peer-connection-createMediaStreamDestination.html is now passing.

Location:

trunk

Files:

• LayoutTests/ChangeLog (modified) (1 diff)
• LayoutTests/platform/gtk/TestExpectations (modified) (1 diff)
• Source/WebCore/ChangeLog (modified) (1 diff)
• Source/WebCore/Modules/webaudio/MediaStreamAudioSource.cpp (modified) (1 diff)
• Source/WebCore/Modules/webaudio/MediaStreamAudioSource.h (modified) (1 diff)
• Source/WebCore/Modules/webaudio/MediaStreamAudioSourceGStreamer.cpp (added)
• Source/WebCore/platform/GStreamer.cmake (modified) (1 diff)
• Source/WebCore/platform/audio/gstreamer/GStreamerAudioData.h (moved) (moved from trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioData.h ) (4 diffs)
• Source/WebCore/platform/audio/gstreamer/GStreamerAudioStreamDescription.h (moved) (moved from trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h ) (4 diffs)
• Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp (modified) (1 diff)

trunk/LayoutTests/ChangeLog

@@ +1 @@
+2020-11-15  Philippe Normand  <pnormand@igalia.com>
+
+        [GStreamer] WebAudio to MediaStream support
+        https://bugs.webkit.org/show_bug.cgi?id=218335
+
+        Reviewed by Xabier Rodriguez-Calvar.
+
+        * platform/gtk/TestExpectations: webrtc/peer-connection-createMediaStreamDestination.html is now passing.
+
 2020-11-14  Rob Buis  <rbuis@igalia.com>
 

trunk/LayoutTests/platform/gtk/TestExpectations

@@ -721 +721 @@
 webkit.org/b/177533 webrtc/video-interruption.html
 
-webkit.org/b/186933 webrtc/peer-connection-createMediaStreamDestination.html
-
 imported/w3c/web-platform-tests/webrtc/ [ Skip ]
 http/tests/webrtc [ Skip ]

trunk/Source/WebCore/ChangeLog

@@ +1 @@
+2020-11-15  Philippe Normand  <pnormand@igalia.com>
+
+        [GStreamer] WebAudio to MediaStream support
+        https://bugs.webkit.org/show_bug.cgi?id=218335
+
+        Reviewed by Xabier Rodriguez-Calvar.
+
+        Provide audio samples coming from the WebAudio bus to the MediaStreamAudioSource node, using
+        GStreamer. This patch also moves a couple files from platform/mediastream to platform/audio,
+        because this is where the virtual parent classes are defined.
+
+        * Modules/webaudio/MediaStreamAudioSource.cpp:
+        * Modules/webaudio/MediaStreamAudioSource.h:
+        * Modules/webaudio/MediaStreamAudioSourceGStreamer.cpp: Added.
+        (WebCore::copyBusData):
+        (WebCore::MediaStreamAudioSource::consumeAudio):
+        * platform/GStreamer.cmake:
+        * platform/audio/gstreamer/GStreamerAudioData.h: Renamed from Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioData.h.
+        (isType):
+        * platform/audio/gstreamer/GStreamerAudioStreamDescription.h: Renamed from Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h.
+        * platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:
+        (WebCore::RealtimeOutgoingAudioSourceLibWebRTC::audioSamplesAvailable):
+        (WebCore::RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData):
+
 2020-11-15  Zalan Bujtas  <zalan@apple.com>
 

trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSource.cpp

@@ -58 +58 @@
 }
 
-#if !PLATFORM(COCOA)
+#if !PLATFORM(COCOA) && !USE(GSTREAMER)
 void MediaStreamAudioSource::consumeAudio(AudioBus&, size_t)
 {

trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSource.h

@@ -61 +61 @@
     RealtimeMediaSourceSettings m_currentSettings;
     std::unique_ptr<PlatformAudioData> m_audioBuffer;
-#if USE(AVFOUNDATION)
+#if USE(AVFOUNDATION) || USE(GSTREAMER)
     size_t m_numberOfFrames { 0 };
 #endif

trunk/Source/WebCore/platform/GStreamer.cmake

@@ -7 +7 @@
 
     list(APPEND WebCore_SOURCES
+        Modules/webaudio/MediaStreamAudioSourceGStreamer.cpp
         platform/graphics/gstreamer/AudioTrackPrivateGStreamer.cpp
         platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp

trunk/Source/WebCore/platform/audio/gstreamer/GStreamerAudioData.h

@@ -22 +22 @@
 #pragma once
 
-#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
+#if USE(GSTREAMER)
 
 #include "GRefPtrGStreamer.h"
@@ -30 +30 @@
 
 namespace WebCore {
+
 class GStreamerAudioData final : public PlatformAudioData {
 public:
@@ -44 +45 @@
     }
 
+    void setSample(GRefPtr<GstSample>&& sample) { m_sample = WTFMove(sample); }
     GstSample* getSample() { return m_sample.get(); }
-    GstAudioInfo getAudioInfo() {return m_audioInfo; }
+    GstAudioInfo getAudioInfo() { return m_audioInfo; }
+    uint32_t channelCount() const { return GST_AUDIO_INFO_CHANNELS(&m_audioInfo); }
 
 private:
     Kind kind() const { return Kind::GStreamerAudioData; }
     GRefPtr<GstSample> m_sample;
-    GRefPtr<GstCaps> m_caps;
-
     GstAudioInfo m_audioInfo;
 };
@@ -57 +58 @@
 } // namespace WebCore
 
-#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
+SPECIALIZE_TYPE_TRAITS_BEGIN(WebCore::GStreamerAudioData)
+static bool isType(const WebCore::PlatformAudioData& data) { return data.kind() == WebCore::PlatformAudioData::Kind::GStreamerAudioData; }
+SPECIALIZE_TYPE_TRAITS_END()
+
+#endif // USE(GSTREAMER)

trunk/Source/WebCore/platform/audio/gstreamer/GStreamerAudioStreamDescription.h

@@ -21 +21 @@
 #pragma once
 
-#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
+#if USE(GSTREAMER)
 
 #include "AudioStreamDescription.h"
@@ -48 +48 @@
     }
 
-    WEBCORE_EXPORT ~GStreamerAudioStreamDescription() { };
+    ~GStreamerAudioStreamDescription() = default;
 
     const PlatformDescription& platformDescription() const
@@ -57 +57 @@
     }
 
-    WEBCORE_EXPORT PCMFormat format() const final {
+    PCMFormat format() const final
+    {
         switch (GST_AUDIO_INFO_FORMAT(&m_info)) {
         case GST_AUDIO_FORMAT_S16LE:
@@ -104 +105 @@
 } // WebCore
 
-#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
+#endif // USE(GSTREAMER)

trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp

@@ -85 +85 @@
     }
 
-    LockHolder locker(m_adapterMutex);
-    auto buffer = gst_sample_get_buffer(data.getSample());
-    gst_adapter_push(m_adapter.get(), gst_buffer_ref(buffer));
+    {
+        LockHolder locker(m_adapterMutex);
+        auto* buffer = gst_sample_get_buffer(data.getSample());
+        gst_adapter_push(m_adapter.get(), gst_buffer_ref(buffer));
+    }
     LibWebRTCProvider::callOnWebRTCSignalingThread([protectedThis = makeRef(*this)] {
         protectedThis->pullAudioData();