Diff
Modified: trunk/LayoutTests/ChangeLog (269826 => 269827)
--- trunk/LayoutTests/ChangeLog 2020-11-15 14:25:42 UTC (rev 269826)
+++ trunk/LayoutTests/ChangeLog 2020-11-15 15:38:57 UTC (rev 269827)
@@ -1,3 +1,12 @@
+2020-11-15 Philippe Normand <[email protected]>
+
+ [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 <[email protected]>
Remove -webkit-aspect-ratio support
Modified: trunk/LayoutTests/platform/gtk/TestExpectations (269826 => 269827)
--- trunk/LayoutTests/platform/gtk/TestExpectations 2020-11-15 14:25:42 UTC (rev 269826)
+++ trunk/LayoutTests/platform/gtk/TestExpectations 2020-11-15 15:38:57 UTC (rev 269827)
@@ -720,8 +720,6 @@
webkit.org/b/187064 webrtc/libwebrtc/descriptionGetters.html
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 ]
Modified: trunk/Source/WebCore/ChangeLog (269826 => 269827)
--- trunk/Source/WebCore/ChangeLog 2020-11-15 14:25:42 UTC (rev 269826)
+++ trunk/Source/WebCore/ChangeLog 2020-11-15 15:38:57 UTC (rev 269827)
@@ -1,3 +1,27 @@
+2020-11-15 Philippe Normand <[email protected]>
+
+ [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 <[email protected]>
[LFC] Do not use RenderStyle's logical margin API
Modified: trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSource.cpp (269826 => 269827)
--- trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSource.cpp 2020-11-15 14:25:42 UTC (rev 269826)
+++ trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSource.cpp 2020-11-15 15:38:57 UTC (rev 269827)
@@ -57,7 +57,7 @@
return m_currentSettings;
}
-#if !PLATFORM(COCOA)
+#if !PLATFORM(COCOA) && !USE(GSTREAMER)
void MediaStreamAudioSource::consumeAudio(AudioBus&, size_t)
{
}
Modified: trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSource.h (269826 => 269827)
--- trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSource.h 2020-11-15 14:25:42 UTC (rev 269826)
+++ trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSource.h 2020-11-15 15:38:57 UTC (rev 269827)
@@ -60,7 +60,7 @@
String m_deviceId;
RealtimeMediaSourceSettings m_currentSettings;
std::unique_ptr<PlatformAudioData> m_audioBuffer;
-#if USE(AVFOUNDATION)
+#if USE(AVFOUNDATION) || USE(GSTREAMER)
size_t m_numberOfFrames { 0 };
#endif
};
Added: trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSourceGStreamer.cpp (0 => 269827)
--- trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSourceGStreamer.cpp (rev 0)
+++ trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSourceGStreamer.cpp 2020-11-15 15:38:57 UTC (rev 269827)
@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2020 Igalia S.L
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public License
+ * aint with this library; see the file COPYING.LIB. If not, write to
+ * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include "config.h"
+#include "MediaStreamAudioSource.h"
+
+#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
+
+#include "AudioBus.h"
+#include "GStreamerAudioData.h"
+#include "GStreamerAudioStreamDescription.h"
+#include "Logging.h"
+
+namespace WebCore {
+
+static void copyBusData(AudioBus& bus, GstBuffer* buffer, bool isMuted)
+{
+ GstMappedBuffer mappedBuffer(buffer, GST_MAP_WRITE);
+ if (isMuted) {
+ memset(mappedBuffer.data(), 0, mappedBuffer.size());
+ return;
+ }
+
+ size_t size = mappedBuffer.size() / bus.numberOfChannels();
+ for (size_t channelIndex = 0; channelIndex < bus.numberOfChannels(); ++channelIndex) {
+ AudioChannel& channel = *bus.channel(channelIndex);
+ float* destination = reinterpret_cast<float*>(mappedBuffer.data() + (channelIndex * size));
+ memcpy(destination, channel.data(), size);
+ }
+}
+
+void MediaStreamAudioSource::consumeAudio(AudioBus& bus, size_t numberOfFrames)
+{
+ if (!bus.numberOfChannels() || bus.numberOfChannels() > 2) {
+ RELEASE_LOG_ERROR(Media, "MediaStreamAudioSource::consumeAudio(%p) trying to consume bus with %u channels", this, bus.numberOfChannels());
+ return;
+ }
+
+ auto mediaTime = MediaTime((m_numberOfFrames * G_USEC_PER_SEC) / m_currentSettings.sampleRate(), G_USEC_PER_SEC);
+ m_numberOfFrames += numberOfFrames;
+
+ GstAudioInfo info;
+ gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_F32LE, m_currentSettings.sampleRate(), bus.numberOfChannels(), nullptr);
+ GST_AUDIO_INFO_LAYOUT(&info) = GST_AUDIO_LAYOUT_NON_INTERLEAVED;
+ size_t size = GST_AUDIO_INFO_BPF(&info) * bus.numberOfChannels() * numberOfFrames;
+
+ auto caps = adoptGRef(gst_audio_info_to_caps(&info));
+ auto buffer = adoptGRef(gst_buffer_new_allocate(nullptr, size, nullptr));
+ copyBusData(bus, buffer.get(), muted());
+ auto sample = adoptGRef(gst_sample_new(buffer.get(), caps.get(), nullptr, nullptr));
+ m_audioBuffer = makeUnique<GStreamerAudioData>(WTFMove(sample), info);
+
+ GStreamerAudioStreamDescription description(info);
+ audioSamplesAvailable(mediaTime, *m_audioBuffer, description, numberOfFrames);
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
Modified: trunk/Source/WebCore/platform/GStreamer.cmake (269826 => 269827)
--- trunk/Source/WebCore/platform/GStreamer.cmake 2020-11-15 14:25:42 UTC (rev 269826)
+++ trunk/Source/WebCore/platform/GStreamer.cmake 2020-11-15 15:38:57 UTC (rev 269827)
@@ -6,6 +6,7 @@
)
list(APPEND WebCore_SOURCES
+ Modules/webaudio/MediaStreamAudioSourceGStreamer.cpp
platform/graphics/gstreamer/AudioTrackPrivateGStreamer.cpp
platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp
platform/graphics/gstreamer/GRefPtrGStreamer.cpp
Copied: trunk/Source/WebCore/platform/audio/gstreamer/GStreamerAudioData.h (from rev 269826, trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioData.h) (0 => 269827)
--- trunk/Source/WebCore/platform/audio/gstreamer/GStreamerAudioData.h (rev 0)
+++ trunk/Source/WebCore/platform/audio/gstreamer/GStreamerAudioData.h 2020-11-15 15:38:57 UTC (rev 269827)
@@ -0,0 +1,64 @@
+/*
+ * Copyright (C) 2018 Metrological Group B.V.
+ * Author: Thibault Saunier <[email protected]>
+ * Author: Alejandro G. Castro <[email protected]>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public License
+ * aint with this library; see the file COPYING.LIB. If not, write to
+ * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#pragma once
+
+#if USE(GSTREAMER)
+
+#include "GRefPtrGStreamer.h"
+#include "PlatformAudioData.h"
+
+#include <gst/audio/audio.h>
+
+namespace WebCore {
+
+class GStreamerAudioData final : public PlatformAudioData {
+public:
+ GStreamerAudioData(GRefPtr<GstSample>&& sample, GstAudioInfo info)
+ : m_sample(WTFMove(sample))
+ , m_audioInfo(info)
+ {
+ }
+
+ GStreamerAudioData(GRefPtr<GstSample>&& sample)
+ : m_sample(WTFMove(sample))
+ {
+ gst_audio_info_from_caps(&m_audioInfo, gst_sample_get_caps(m_sample.get()));
+ }
+
+ void setSample(GRefPtr<GstSample>&& sample) { m_sample = WTFMove(sample); }
+ GstSample* getSample() { return m_sample.get(); }
+ 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;
+ GstAudioInfo m_audioInfo;
+};
+
+} // namespace WebCore
+
+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)
Copied: trunk/Source/WebCore/platform/audio/gstreamer/GStreamerAudioStreamDescription.h (from rev 269826, trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h) (0 => 269827)
--- trunk/Source/WebCore/platform/audio/gstreamer/GStreamerAudioStreamDescription.h (rev 0)
+++ trunk/Source/WebCore/platform/audio/gstreamer/GStreamerAudioStreamDescription.h 2020-11-15 15:38:57 UTC (rev 269827)
@@ -0,0 +1,107 @@
+/*
+ * Copyright (C) 2018 Metrological Group B.V.
+ * Copyright (C) 2018, 2020 Igalia S.L.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public License
+ * aint with this library; see the file COPYING.LIB. If not, write to
+ * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#pragma once
+
+#if USE(GSTREAMER)
+
+#include "AudioStreamDescription.h"
+#include "GStreamerCommon.h"
+#include <gst/audio/audio.h>
+
+namespace WebCore {
+
+class GStreamerAudioStreamDescription final: public AudioStreamDescription {
+public:
+ GStreamerAudioStreamDescription(GstAudioInfo info)
+ : m_info(info)
+ , m_caps(adoptGRef(gst_audio_info_to_caps(&m_info)))
+ {
+ }
+
+ GStreamerAudioStreamDescription(GstAudioInfo *info)
+ : m_info(*info)
+ , m_caps(adoptGRef(gst_audio_info_to_caps(&m_info)))
+ {
+ }
+
+ GStreamerAudioStreamDescription()
+ {
+ gst_audio_info_init(&m_info);
+ }
+
+ ~GStreamerAudioStreamDescription() = default;
+
+ const PlatformDescription& platformDescription() const
+ {
+ m_platformDescription = { PlatformDescription::GStreamerAudioStreamDescription, reinterpret_cast<const AudioStreamBasicDescription*>(&m_info) };
+
+ return m_platformDescription;
+ }
+
+ PCMFormat format() const final
+ {
+ switch (GST_AUDIO_INFO_FORMAT(&m_info)) {
+ case GST_AUDIO_FORMAT_S16LE:
+ case GST_AUDIO_FORMAT_S16BE:
+ return Int16;
+ case GST_AUDIO_FORMAT_S32LE:
+ case GST_AUDIO_FORMAT_S32BE:
+ return Int32;
+ case GST_AUDIO_FORMAT_F32LE:
+ case GST_AUDIO_FORMAT_F32BE:
+ return Float32;
+ case GST_AUDIO_FORMAT_F64LE:
+ case GST_AUDIO_FORMAT_F64BE:
+ return Float64;
+ default:
+ break;
+ }
+ return None;
+ }
+
+ double sampleRate() const final { return GST_AUDIO_INFO_RATE(&m_info); }
+ bool isPCM() const final { return format() != None; }
+ bool isInterleaved() const final { return GST_AUDIO_INFO_LAYOUT(&m_info) == GST_AUDIO_LAYOUT_INTERLEAVED; }
+ bool isSignedInteger() const final { return GST_AUDIO_INFO_IS_INTEGER(&m_info); }
+ bool isNativeEndian() const final { return GST_AUDIO_INFO_ENDIANNESS(&m_info) == G_BYTE_ORDER; }
+ bool isFloat() const final { return GST_AUDIO_INFO_IS_FLOAT(&m_info); }
+ int bytesPerFrame() { return GST_AUDIO_INFO_BPF(&m_info); }
+
+ uint32_t numberOfInterleavedChannels() const final { return isInterleaved() ? GST_AUDIO_INFO_CHANNELS(&m_info) : TRUE; }
+ uint32_t numberOfChannelStreams() const final { return GST_AUDIO_INFO_CHANNELS(&m_info); }
+ uint32_t numberOfChannels() const final { return GST_AUDIO_INFO_CHANNELS(&m_info); }
+ uint32_t sampleWordSize() const final { return GST_AUDIO_INFO_BPS(&m_info); }
+
+ bool operator==(const GStreamerAudioStreamDescription& other) { return gst_audio_info_is_equal(&m_info, &other.m_info); }
+ bool operator!=(const GStreamerAudioStreamDescription& other) { return !operator == (other); }
+
+ GstCaps* caps() { return m_caps.get(); }
+ GstAudioInfo* getInfo() { return &m_info; }
+
+private:
+ GstAudioInfo m_info;
+ GRefPtr<GstCaps> m_caps;
+ mutable PlatformDescription m_platformDescription;
+};
+
+} // WebCore
+
+#endif // USE(GSTREAMER)
Deleted: trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioData.h (269826 => 269827)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioData.h 2020-11-15 14:25:42 UTC (rev 269826)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioData.h 2020-11-15 15:38:57 UTC (rev 269827)
@@ -1,59 +0,0 @@
-/*
- * Copyright (C) 2018 Metrological Group B.V.
- * Author: Thibault Saunier <[email protected]>
- * Author: Alejandro G. Castro <[email protected]>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public License
- * aint with this library; see the file COPYING.LIB. If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#pragma once
-
-#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
-
-#include "GRefPtrGStreamer.h"
-#include "PlatformAudioData.h"
-
-#include <gst/audio/audio.h>
-
-namespace WebCore {
-class GStreamerAudioData final : public PlatformAudioData {
-public:
- GStreamerAudioData(GRefPtr<GstSample>&& sample, GstAudioInfo info)
- : m_sample(WTFMove(sample))
- , m_audioInfo(info)
- {
- }
-
- GStreamerAudioData(GRefPtr<GstSample>&& sample)
- : m_sample(WTFMove(sample))
- {
- gst_audio_info_from_caps(&m_audioInfo, gst_sample_get_caps(m_sample.get()));
- }
-
- GstSample* getSample() { return m_sample.get(); }
- GstAudioInfo getAudioInfo() {return m_audioInfo; }
-
-private:
- Kind kind() const { return Kind::GStreamerAudioData; }
- GRefPtr<GstSample> m_sample;
- GRefPtr<GstCaps> m_caps;
-
- GstAudioInfo m_audioInfo;
-};
-
-} // namespace WebCore
-
-#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
Deleted: trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h (269826 => 269827)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h 2020-11-15 14:25:42 UTC (rev 269826)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h 2020-11-15 15:38:57 UTC (rev 269827)
@@ -1,106 +0,0 @@
-/*
- * Copyright (C) 2018 Metrological Group B.V.
- * Copyright (C) 2018, 2020 Igalia S.L.
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public License
- * aint with this library; see the file COPYING.LIB. If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#pragma once
-
-#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
-
-#include "AudioStreamDescription.h"
-#include "GStreamerCommon.h"
-#include <gst/audio/audio.h>
-
-namespace WebCore {
-
-class GStreamerAudioStreamDescription final: public AudioStreamDescription {
-public:
- GStreamerAudioStreamDescription(GstAudioInfo info)
- : m_info(info)
- , m_caps(adoptGRef(gst_audio_info_to_caps(&m_info)))
- {
- }
-
- GStreamerAudioStreamDescription(GstAudioInfo *info)
- : m_info(*info)
- , m_caps(adoptGRef(gst_audio_info_to_caps(&m_info)))
- {
- }
-
- GStreamerAudioStreamDescription()
- {
- gst_audio_info_init(&m_info);
- }
-
- WEBCORE_EXPORT ~GStreamerAudioStreamDescription() { };
-
- const PlatformDescription& platformDescription() const
- {
- m_platformDescription = { PlatformDescription::GStreamerAudioStreamDescription, reinterpret_cast<const AudioStreamBasicDescription*>(&m_info) };
-
- return m_platformDescription;
- }
-
- WEBCORE_EXPORT PCMFormat format() const final {
- switch (GST_AUDIO_INFO_FORMAT(&m_info)) {
- case GST_AUDIO_FORMAT_S16LE:
- case GST_AUDIO_FORMAT_S16BE:
- return Int16;
- case GST_AUDIO_FORMAT_S32LE:
- case GST_AUDIO_FORMAT_S32BE:
- return Int32;
- case GST_AUDIO_FORMAT_F32LE:
- case GST_AUDIO_FORMAT_F32BE:
- return Float32;
- case GST_AUDIO_FORMAT_F64LE:
- case GST_AUDIO_FORMAT_F64BE:
- return Float64;
- default:
- break;
- }
- return None;
- }
-
- double sampleRate() const final { return GST_AUDIO_INFO_RATE(&m_info); }
- bool isPCM() const final { return format() != None; }
- bool isInterleaved() const final { return GST_AUDIO_INFO_LAYOUT(&m_info) == GST_AUDIO_LAYOUT_INTERLEAVED; }
- bool isSignedInteger() const final { return GST_AUDIO_INFO_IS_INTEGER(&m_info); }
- bool isNativeEndian() const final { return GST_AUDIO_INFO_ENDIANNESS(&m_info) == G_BYTE_ORDER; }
- bool isFloat() const final { return GST_AUDIO_INFO_IS_FLOAT(&m_info); }
- int bytesPerFrame() { return GST_AUDIO_INFO_BPF(&m_info); }
-
- uint32_t numberOfInterleavedChannels() const final { return isInterleaved() ? GST_AUDIO_INFO_CHANNELS(&m_info) : TRUE; }
- uint32_t numberOfChannelStreams() const final { return GST_AUDIO_INFO_CHANNELS(&m_info); }
- uint32_t numberOfChannels() const final { return GST_AUDIO_INFO_CHANNELS(&m_info); }
- uint32_t sampleWordSize() const final { return GST_AUDIO_INFO_BPS(&m_info); }
-
- bool operator==(const GStreamerAudioStreamDescription& other) { return gst_audio_info_is_equal(&m_info, &other.m_info); }
- bool operator!=(const GStreamerAudioStreamDescription& other) { return !operator == (other); }
-
- GstCaps* caps() { return m_caps.get(); }
- GstAudioInfo* getInfo() { return &m_info; }
-
-private:
- GstAudioInfo m_info;
- GRefPtr<GstCaps> m_caps;
- mutable PlatformDescription m_platformDescription;
-};
-
-} // WebCore
-
-#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp (269826 => 269827)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp 2020-11-15 14:25:42 UTC (rev 269826)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp 2020-11-15 15:38:57 UTC (rev 269827)
@@ -84,9 +84,11 @@
nullptr));
}
- 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();
});
