Diff
Modified: trunk/Source/WebCore/CMakeLists.txt (223411 => 223412)
--- trunk/Source/WebCore/CMakeLists.txt 2017-10-16 16:38:15 UTC (rev 223411)
+++ trunk/Source/WebCore/CMakeLists.txt 2017-10-16 16:40:48 UTC (rev 223412)
@@ -2539,10 +2539,12 @@
platform/mediastream/RealtimeMediaSourceSupportedConstraints.cpp
platform/mediastream/RTCIceCandidateDescriptor.cpp
platform/mediastream/RTCSessionDescriptionDescriptor.cpp
+ platform/mediastream/RealtimeIncomingAudioSource.cpp
platform/mediastream/RealtimeIncomingVideoSource.cpp
platform/mediastream/RealtimeMediaSource.cpp
platform/mediastream/RealtimeMediaSourceCenter.cpp
platform/mediastream/RealtimeMediaSourceSettings.cpp
+ platform/mediastream/RealtimeOutgoingAudioSource.cpp
platform/mediastream/RealtimeOutgoingVideoSource.cpp
platform/mediastream/libwebrtc/LibWebRTCProvider.cpp
Modified: trunk/Source/WebCore/ChangeLog (223411 => 223412)
--- trunk/Source/WebCore/ChangeLog 2017-10-16 16:38:15 UTC (rev 223411)
+++ trunk/Source/WebCore/ChangeLog 2017-10-16 16:40:48 UTC (rev 223412)
@@ -1,3 +1,29 @@
+2017-10-16 Alejandro G. Castro <[email protected]>
+
+ Make RealtimeIncomingAudioSources and RealtimeOutgoingAudioSources port agnostic
+ https://bugs.webkit.org/show_bug.cgi?id=177928
+
+ Reviewed by Youenn Fablet.
+
+ Refactor the RealtimeIncomingAudioSource and RealtimeOutgoingAudioSource classes,
+ move the mac specific code to a different class. This way we can use them from
+ other ports.
+
+ No new tests, we are not adding new functionality just refactoring.
+
+ * CMakeLists.txt:
+ * WebCore.xcodeproj/project.pbxproj:
+ * platform/mediastream/RealtimeIncomingAudioSource.cpp: Copied from Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h. Just the port agnostic parts.
+ * platform/mediastream/RealtimeIncomingAudioSource.h: Copied from Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h. Ditto.
+ * platform/mediastream/RealtimeOutgoingAudioSource.cpp: Copied from Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp. Ditto.
+ * platform/mediastream/RealtimeOutgoingAudioSource.h: Copied from Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h. Ditto.
+ * platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.cpp: Added, it adds Cocoa code parts from original Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp.
+ (WebCore::RealtimeIncomingAudioSource::create): Moved this function from the RealtimeOutgoingAudioSource.cpp file to avoid including the Cocoa file there.
+ * platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.h: Added Cocoa code from original Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h.
+ * platform/mediastream/mac/RealtimeOutgoingAudioSourceCocoa.cpp: Added Cocoa code from original Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp.
+ (WebCore::RealtimeOutgoingAudioSource::create): Moved this function from the RealtimeOutgoingAudioSource.cpp file to avoid including the Cocoa file there.
+ * platform/mediastream/mac/RealtimeOutgoingAudioSourceCocoa.h: Added Cocoa code from the original Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h.
+
2017-10-16 Wenson Hsieh <[email protected]>
Unreviewed, fix the tvOS build after r223340.
Modified: trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj (223411 => 223412)
--- trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2017-10-16 16:38:15 UTC (rev 223411)
+++ trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2017-10-16 16:40:48 UTC (rev 223412)
@@ -1675,8 +1675,12 @@
410B7E721045FAB000D8224F /* JSMessageEventCustom.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 410B7E711045FAB000D8224F /* JSMessageEventCustom.cpp */; };
41103AAB1E39791000769F03 /* RealtimeOutgoingAudioSource.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 41103AA71E39790A00769F03 /* RealtimeOutgoingAudioSource.cpp */; };
41103AAC1E39791000769F03 /* RealtimeOutgoingAudioSource.h in Headers */ = {isa = PBXBuildFile; fileRef = 41103AA81E39790A00769F03 /* RealtimeOutgoingAudioSource.h */; };
+ 41103AAB1E39791000769F14 /* RealtimeOutgoingAudioSourceCocoa.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 41103AA71E39790A00769F14 /* RealtimeOutgoingAudioSourceCocoa.cpp */; };
+ 41103AAC1E39791000769F14 /* RealtimeOutgoingAudioSourceCocoa.h in Headers */ = {isa = PBXBuildFile; fileRef = 41103AA81E39790A00769F14 /* RealtimeOutgoingAudioSourceCocoa.h */; };
41103AAD1E39791000769F03 /* RealtimeIncomingAudioSource.h in Headers */ = {isa = PBXBuildFile; fileRef = 41103AA91E39790A00769F03 /* RealtimeIncomingAudioSource.h */; };
41103AAE1E39791000769F03 /* RealtimeIncomingAudioSource.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 41103AAA1E39790A00769F03 /* RealtimeIncomingAudioSource.cpp */; };
+ 41103AAD1E39791000769F14 /* RealtimeIncomingAudioSourceCocoa.h in Headers */ = {isa = PBXBuildFile; fileRef = 41103AA91E39790A00769F14 /* RealtimeIncomingAudioSourceCocoa.h */; };
+ 41103AAE1E39791000769F14 /* RealtimeIncomingAudioSourceCocoa.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 41103AAA1E39790A00769F14 /* RealtimeIncomingAudioSourceCocoa.cpp */; };
4123081B138C429700BCCFCA /* WebCore.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 93F19B1A08245E5A001E9ABC /* WebCore.framework */; };
41230913138C42FF00BCCFCA /* _javascript_Core.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = F8216299029F4FB501000131 /* _javascript_Core.framework */; };
4127D5370F8AAB1D00E424F5 /* ScriptState.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 4127D5360F8AAB1D00E424F5 /* ScriptState.cpp */; };
@@ -9473,8 +9477,12 @@
410B7E711045FAB000D8224F /* JSMessageEventCustom.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSMessageEventCustom.cpp; sourceTree = "<group>"; };
41103AA71E39790A00769F03 /* RealtimeOutgoingAudioSource.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = RealtimeOutgoingAudioSource.cpp; sourceTree = "<group>"; };
41103AA81E39790A00769F03 /* RealtimeOutgoingAudioSource.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RealtimeOutgoingAudioSource.h; sourceTree = "<group>"; };
+ 41103AA71E39790A00769F14 /* RealtimeOutgoingAudioSourceCocoa.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = RealtimeOutgoingAudioSourceCocoa.cpp; sourceTree = "<group>"; };
+ 41103AA81E39790A00769F14 /* RealtimeOutgoingAudioSourceCocoa.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RealtimeOutgoingAudioSourceCocoa.h; sourceTree = "<group>"; };
41103AA91E39790A00769F03 /* RealtimeIncomingAudioSource.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RealtimeIncomingAudioSource.h; sourceTree = "<group>"; };
41103AAA1E39790A00769F03 /* RealtimeIncomingAudioSource.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = RealtimeIncomingAudioSource.cpp; sourceTree = "<group>"; };
+ 41103AA91E39790A00769F14 /* RealtimeIncomingAudioSourceCocoa.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RealtimeIncomingAudioSourceCocoa.h; sourceTree = "<group>"; };
+ 41103AAA1E39790A00769F14 /* RealtimeIncomingAudioSourceCocoa.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = RealtimeIncomingAudioSourceCocoa.cpp; sourceTree = "<group>"; };
41189EF71AD8232800B90A0D /* ReadableStreamDefaultController.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = ReadableStreamDefaultController.idl; sourceTree = "<group>"; };
41189EF71AD8232800B93F64 /* ReadableByteStreamController.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = ReadableByteStreamController.idl; sourceTree = "<group>"; };
41189EF71AD8232800B95672 /* ReadableStreamBYOBRequest.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = ReadableStreamBYOBRequest.idl; sourceTree = "<group>"; };
@@ -16564,6 +16572,8 @@
07FFDE66181AED420072D409 /* MediaStreamTrackPrivate.cpp */,
07FFDE67181AED420072D409 /* MediaStreamTrackPrivate.h */,
5EBB89381C77BDA400C65D41 /* PeerMediaDescription.h */,
+ 41103AAA1E39790A00769F03 /* RealtimeIncomingAudioSource.cpp */,
+ 41103AA91E39790A00769F03 /* RealtimeIncomingAudioSource.h */,
5CDD83391E4324BB00621E92 /* RealtimeIncomingVideoSource.cpp */,
5CDD833A1E4324BB00621E92 /* RealtimeIncomingVideoSource.h */,
4A4F656B1AA997F100E38CDD /* RealtimeMediaSource.cpp */,
@@ -16575,6 +16585,8 @@
4A4F656F1AA997F100E38CDD /* RealtimeMediaSourceSettings.h */,
2EC41DE21C0410A300D294FE /* RealtimeMediaSourceSupportedConstraints.cpp */,
07C1C0E41BFB60ED00BD2256 /* RealtimeMediaSourceSupportedConstraints.h */,
+ 41103AA71E39790A00769F03 /* RealtimeOutgoingAudioSource.cpp */,
+ 41103AA81E39790A00769F03 /* RealtimeOutgoingAudioSource.h */,
5CDD833B1E4324BB00621E92 /* RealtimeOutgoingVideoSource.cpp */,
5CDD833C1E4324BB00621E92 /* RealtimeOutgoingVideoSource.h */,
3135910C1E7DDCB600F30630 /* RTCBundlePolicy.h */,
@@ -16627,6 +16639,8 @@
07EE76EE1BEA619800F89133 /* MockRealtimeVideoSourceMac.mm */,
41103AAA1E39790A00769F03 /* RealtimeIncomingAudioSource.cpp */,
41103AA91E39790A00769F03 /* RealtimeIncomingAudioSource.h */,
+ 41103AAA1E39790A00769F14 /* RealtimeIncomingAudioSourceCocoa.cpp */,
+ 41103AA91E39790A00769F14 /* RealtimeIncomingAudioSourceCocoa.h */,
5CDD83391E4324BB00621E83 /* RealtimeIncomingVideoSourceCocoa.cpp */,
5CDD833A1E4324BB00621E83 /* RealtimeIncomingVideoSourceCocoa.h */,
4A0FFAA31AAF5EF60062803B /* RealtimeMediaSourceCenterMac.cpp */,
@@ -16633,6 +16647,8 @@
4A0FFAA41AAF5EF60062803B /* RealtimeMediaSourceCenterMac.h */,
41103AA71E39790A00769F03 /* RealtimeOutgoingAudioSource.cpp */,
41103AA81E39790A00769F03 /* RealtimeOutgoingAudioSource.h */,
+ 41103AA71E39790A00769F14 /* RealtimeOutgoingAudioSourceCocoa.cpp */,
+ 41103AA81E39790A00769F14 /* RealtimeOutgoingAudioSourceCocoa.h */,
5CDD833B1E4324BB00621B83 /* RealtimeOutgoingVideoSourceCocoa.cpp */,
5CDD833C1E4324BB00621B83 /* RealtimeOutgoingVideoSourceCocoa.h */,
07D6373E1BB0B11300256CE9 /* WebAudioSourceProviderAVFObjC.h */,
@@ -29772,6 +29788,7 @@
4129C9AB1F59C573009D7403 /* ReadableStreamSource.h in Headers */,
FD31603C12B0267600C1A359 /* RealtimeAnalyser.h in Headers */,
41103AAD1E39791000769F03 /* RealtimeIncomingAudioSource.h in Headers */,
+ 41103AAD1E39791000769F14 /* RealtimeIncomingAudioSourceCocoa.h in Headers */,
4A4F65711AA997F100E38CDD /* RealtimeMediaSource.h in Headers */,
4A4F65721AA997F100E38CDD /* RealtimeMediaSourceCapabilities.h in Headers */,
4A0FFAA21AAF5EA20062803B /* RealtimeMediaSourceCenter.h in Headers */,
@@ -29779,6 +29796,7 @@
4A4F65741AA997F100E38CDD /* RealtimeMediaSourceSettings.h in Headers */,
07C1C0E51BFB60ED00BD2256 /* RealtimeMediaSourceSupportedConstraints.h in Headers */,
41103AAC1E39791000769F03 /* RealtimeOutgoingAudioSource.h in Headers */,
+ 41103AAC1E39791000769F14 /* RealtimeOutgoingAudioSourceCocoa.h in Headers */,
91B952241F58A58F00931DC2 /* RecordingSwizzleTypes.h in Headers */,
BC4368E80C226E32005EFB5F /* Rect.h in Headers */,
FD45A958175D414C00C21EC8 /* RectangleShape.h in Headers */,
@@ -33752,6 +33770,7 @@
4129C98A1F587FEB009D7403 /* ReadableStreamSink.cpp in Sources */,
FD31603B12B0267600C1A359 /* RealtimeAnalyser.cpp in Sources */,
41103AAE1E39791000769F03 /* RealtimeIncomingAudioSource.cpp in Sources */,
+ 41103AAE1E39791000769F14 /* RealtimeIncomingAudioSourceCocoa.cpp in Sources */,
5CDD833E1E4324DC00621E92 /* RealtimeIncomingVideoSource.cpp in Sources */,
5CDD833E1E4324DC00621E83 /* RealtimeIncomingVideoSourceCocoa.cpp in Sources */,
4A4F65701AA997F100E38CDD /* RealtimeMediaSource.cpp in Sources */,
@@ -33760,6 +33779,7 @@
4A4F65731AA997F100E38CDD /* RealtimeMediaSourceSettings.cpp in Sources */,
2EC41DE41C0410A300D294FE /* RealtimeMediaSourceSupportedConstraints.cpp in Sources */,
41103AAB1E39791000769F03 /* RealtimeOutgoingAudioSource.cpp in Sources */,
+ 41103AAB1E39791000769F14 /* RealtimeOutgoingAudioSourceCocoa.cpp in Sources */,
5CDD833D1E4324D900621E92 /* RealtimeOutgoingVideoSource.cpp in Sources */,
5CDD833D1E4324D900621B83 /* RealtimeOutgoingVideoSourceCocoa.cpp in Sources */,
FD45A95A175D417100C21EC8 /* RectangleShape.cpp in Sources */,
Copied: trunk/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.cpp (from rev 223411, trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h) (0 => 223412)
--- trunk/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.cpp (rev 0)
+++ trunk/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.cpp 2017-10-16 16:40:48 UTC (rev 223412)
@@ -0,0 +1,87 @@
+/*
+ * Copyright (C) 2017 Apple Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer
+ * in the documentation and/or other materials provided with the
+ * distribution.
+ * 3. Neither the name of Google Inc. nor the names of its contributors
+ * may be used to endorse or promote products derived from this
+ * software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "config.h"
+#include "RealtimeIncomingAudioSource.h"
+
+#if USE(LIBWEBRTC)
+
+#include "LibWebRTCAudioFormat.h"
+
+namespace WebCore {
+
+RealtimeIncomingAudioSource::RealtimeIncomingAudioSource(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
+ : RealtimeMediaSource(WTFMove(audioTrackId), RealtimeMediaSource::Type::Audio, String())
+ , m_audioTrack(WTFMove(audioTrack))
+{
+ notifyMutedChange(!m_audioTrack);
+}
+
+RealtimeIncomingAudioSource::~RealtimeIncomingAudioSource()
+{
+ stop();
+}
+
+void RealtimeIncomingAudioSource::startProducingData()
+{
+ if (m_audioTrack)
+ m_audioTrack->AddSink(this);
+}
+
+void RealtimeIncomingAudioSource::stopProducingData()
+{
+ if (m_audioTrack)
+ m_audioTrack->RemoveSink(this);
+}
+
+void RealtimeIncomingAudioSource::setSourceTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& track)
+{
+ ASSERT(!m_audioTrack);
+ ASSERT(track);
+
+ m_audioTrack = WTFMove(track);
+ notifyMutedChange(!m_audioTrack);
+ if (isProducingData())
+ m_audioTrack->AddSink(this);
+}
+
+const RealtimeMediaSourceCapabilities& RealtimeIncomingAudioSource::capabilities() const
+{
+ return RealtimeMediaSourceCapabilities::emptyCapabilities();
+}
+
+const RealtimeMediaSourceSettings& RealtimeIncomingAudioSource::settings() const
+{
+ return m_currentSettings;
+}
+
+}
+
+#endif // USE(LIBWEBRTC)
Copied: trunk/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.h (from rev 223411, trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h) (0 => 223412)
--- trunk/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.h (rev 0)
+++ trunk/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.h 2017-10-16 16:40:48 UTC (rev 223412)
@@ -0,0 +1,69 @@
+/*
+ * Copyright (C) 2017 Apple Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer
+ * in the documentation and/or other materials provided with the
+ * distribution.
+ * 3. Neither the name of Ericsson nor the names of its contributors
+ * may be used to endorse or promote products derived from this
+ * software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#pragma once
+
+#if USE(LIBWEBRTC)
+
+#include "LibWebRTCMacros.h"
+#include "RealtimeMediaSource.h"
+#include <webrtc/api/mediastreaminterface.h>
+#include <wtf/RetainPtr.h>
+
+namespace WebCore {
+
+class RealtimeIncomingAudioSource : public RealtimeMediaSource, private webrtc::AudioTrackSinkInterface {
+public:
+ static Ref<RealtimeIncomingAudioSource> create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&&, String&&);
+
+ void setSourceTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>&&);
+
+protected:
+ RealtimeIncomingAudioSource(rtc::scoped_refptr<webrtc::AudioTrackInterface>&&, String&&);
+ ~RealtimeIncomingAudioSource();
+
+private:
+ // webrtc::AudioTrackSinkInterface API
+ virtual void OnData(const void* /* audioData */, int /* bitsPerSample */, int /* sampleRate */, size_t /* numberOfChannels */, size_t /* numberOfFrames */) { };
+
+ // RealtimeMediaSource API
+ void startProducingData() final;
+ void stopProducingData() final;
+
+ const RealtimeMediaSourceCapabilities& capabilities() const final;
+ const RealtimeMediaSourceSettings& settings() const final;
+
+ RealtimeMediaSourceSettings m_currentSettings;
+ rtc::scoped_refptr<webrtc::AudioTrackInterface> m_audioTrack;
+};
+
+} // namespace WebCore
+
+#endif // USE(LIBWEBRTC)
Added: trunk/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.cpp (0 => 223412)
--- trunk/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.cpp (rev 0)
+++ trunk/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.cpp 2017-10-16 16:40:48 UTC (rev 223412)
@@ -0,0 +1,93 @@
+/*
+ * Copyright (C) 2017 Apple Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted, provided that the following conditions
+ * are required to be met:
+ *
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ * 3. Neither the name of Apple Inc. nor the names of
+ * its contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+ * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "config.h"
+#include "RealtimeOutgoingAudioSource.h"
+
+#if USE(LIBWEBRTC)
+
+#include "LibWebRTCAudioFormat.h"
+#include "LibWebRTCProvider.h"
+
+namespace WebCore {
+
+RealtimeOutgoingAudioSource::RealtimeOutgoingAudioSource(Ref<MediaStreamTrackPrivate>&& audioSource)
+ : m_audioSource(WTFMove(audioSource))
+ , m_silenceAudioTimer(*this, &RealtimeOutgoingAudioSource::sendSilence)
+{
+ m_audioSource->addObserver(*this);
+ initializeConverter();
+}
+
+bool RealtimeOutgoingAudioSource::setSource(Ref<MediaStreamTrackPrivate>&& newSource)
+{
+ m_audioSource->removeObserver(*this);
+ m_audioSource = WTFMove(newSource);
+ m_audioSource->addObserver(*this);
+
+ initializeConverter();
+ return true;
+}
+
+void RealtimeOutgoingAudioSource::initializeConverter()
+{
+ m_muted = m_audioSource->muted();
+ m_enabled = m_audioSource->enabled();
+ handleMutedIfNeeded();
+}
+
+void RealtimeOutgoingAudioSource::stop()
+{
+ m_silenceAudioTimer.stop();
+ m_audioSource->removeObserver(*this);
+}
+
+void RealtimeOutgoingAudioSource::sourceMutedChanged()
+{
+ m_muted = m_audioSource->muted();
+ handleMutedIfNeeded();
+}
+
+void RealtimeOutgoingAudioSource::sourceEnabledChanged()
+{
+ m_enabled = m_audioSource->enabled();
+ handleMutedIfNeeded();
+}
+
+void RealtimeOutgoingAudioSource::handleMutedIfNeeded()
+{
+ bool isSilenced = m_muted || !m_enabled;
+ if (isSilenced && !m_silenceAudioTimer.isActive())
+ m_silenceAudioTimer.startRepeating(1_s);
+ if (!isSilenced && m_silenceAudioTimer.isActive())
+ m_silenceAudioTimer.stop();
+}
+
+} // namespace WebCore
+
+#endif // USE(LIBWEBRTC)
Copied: trunk/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h (from rev 223411, trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h) (0 => 223412)
--- trunk/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h (rev 0)
+++ trunk/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h 2017-10-16 16:40:48 UTC (rev 223412)
@@ -0,0 +1,104 @@
+/*
+ * Copyright (C) 2017 Apple Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted, provided that the following conditions
+ * are required to be met:
+ *
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ * 3. Neither the name of Apple Inc. nor the names of
+ * its contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+ * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#pragma once
+
+
+#if USE(LIBWEBRTC)
+
+#include "LibWebRTCMacros.h"
+#include "MediaStreamTrackPrivate.h"
+#include "Timer.h"
+#include <webrtc/api/mediastreaminterface.h>
+#include <wtf/ThreadSafeRefCounted.h>
+
+namespace webrtc {
+class AudioTrackInterface;
+class AudioTrackSinkInterface;
+}
+
+namespace WebCore {
+
+class RealtimeOutgoingAudioSource : public ThreadSafeRefCounted<RealtimeOutgoingAudioSource>, public webrtc::AudioSourceInterface, private MediaStreamTrackPrivate::Observer {
+public:
+ static Ref<RealtimeOutgoingAudioSource> create(Ref<MediaStreamTrackPrivate>&& audioSource);
+
+ ~RealtimeOutgoingAudioSource() { stop(); }
+
+ void stop();
+
+ bool setSource(Ref<MediaStreamTrackPrivate>&&);
+ MediaStreamTrackPrivate& source() const { return m_audioSource.get(); }
+
+protected:
+ explicit RealtimeOutgoingAudioSource(Ref<MediaStreamTrackPrivate>&&);
+
+ virtual void handleMutedIfNeeded();
+ virtual void sendSilence() { };
+ virtual void pullAudioData() { };
+
+ Vector<webrtc::AudioTrackSinkInterface*> m_sinks;
+ bool m_muted { false };
+ bool m_enabled { true };
+
+private:
+ virtual void AddSink(webrtc::AudioTrackSinkInterface* sink) { m_sinks.append(sink); }
+ virtual void RemoveSink(webrtc::AudioTrackSinkInterface* sink) { m_sinks.removeFirst(sink); }
+
+ int AddRef() const final { ref(); return refCount(); }
+ int Release() const final { deref(); return refCount(); }
+ SourceState state() const final { return kLive; }
+ bool remote() const final { return false; }
+ void RegisterObserver(webrtc::ObserverInterface*) final { }
+ void UnregisterObserver(webrtc::ObserverInterface*) final { }
+
+ void sourceMutedChanged();
+ void sourceEnabledChanged();
+ virtual void audioSamplesAvailable(const MediaTime&, const PlatformAudioData&, const AudioStreamDescription&, size_t) { };
+
+ virtual bool isReachingBufferedAudioDataHighLimit() { return false; };
+ virtual bool isReachingBufferedAudioDataLowLimit() { return false; };
+ virtual bool hasBufferedEnoughData() { return false; };
+
+ // MediaStreamTrackPrivate::Observer API
+ void trackMutedChanged(MediaStreamTrackPrivate&) final { sourceMutedChanged(); }
+ void trackEnabledChanged(MediaStreamTrackPrivate&) final { sourceEnabledChanged(); }
+ void audioSamplesAvailable(MediaStreamTrackPrivate&, const MediaTime& mediaTime, const PlatformAudioData& data, const AudioStreamDescription& description, size_t sampleCount) { audioSamplesAvailable(mediaTime, data, description, sampleCount); }
+ void trackEnded(MediaStreamTrackPrivate&) final { }
+ void trackSettingsChanged(MediaStreamTrackPrivate&) final { }
+
+ void initializeConverter();
+
+ Ref<MediaStreamTrackPrivate> m_audioSource;
+
+ Timer m_silenceAudioTimer;
+};
+
+} // namespace WebCore
+
+#endif // USE(LIBWEBRTC)
Deleted: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp (223411 => 223412)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp 2017-10-16 16:38:15 UTC (rev 223411)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp 2017-10-16 16:40:48 UTC (rev 223412)
@@ -1,140 +0,0 @@
-/*
- * Copyright (C) 2017 Apple Inc. All rights reserved.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions
- * are met:
- *
- * 1. Redistributions of source code must retain the above copyright
- * notice, this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright
- * notice, this list of conditions and the following disclaimer
- * in the documentation and/or other materials provided with the
- * distribution.
- * 3. Neither the name of Google Inc. nor the names of its contributors
- * may be used to endorse or promote products derived from this
- * software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
- * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
- * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
- * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
- * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
- * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
- * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
- * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
- * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
- * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "config.h"
-#include "RealtimeIncomingAudioSource.h"
-
-#if USE(LIBWEBRTC)
-
-#include "AudioStreamDescription.h"
-#include "CAAudioStreamDescription.h"
-#include "LibWebRTCAudioFormat.h"
-#include "WebAudioBufferList.h"
-#include "WebAudioSourceProviderAVFObjC.h"
-#include <pal/avfoundation/MediaTimeAVFoundation.h>
-
-#include "CoreMediaSoftLink.h"
-
-namespace WebCore {
-
-Ref<RealtimeIncomingAudioSource> RealtimeIncomingAudioSource::create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
-{
- auto source = adoptRef(*new RealtimeIncomingAudioSource(WTFMove(audioTrack), WTFMove(audioTrackId)));
- source->start();
- return source;
-}
-
-RealtimeIncomingAudioSource::RealtimeIncomingAudioSource(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
- : RealtimeMediaSource(WTFMove(audioTrackId), RealtimeMediaSource::Type::Audio, String())
- , m_audioTrack(WTFMove(audioTrack))
-{
- notifyMutedChange(!m_audioTrack);
-}
-
-RealtimeIncomingAudioSource::~RealtimeIncomingAudioSource()
-{
- stop();
-}
-
-
-static inline AudioStreamBasicDescription streamDescription(size_t sampleRate, size_t channelCount)
-{
- AudioStreamBasicDescription streamFormat;
- FillOutASBDForLPCM(streamFormat, sampleRate, channelCount, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::isFloat, LibWebRTCAudioFormat::isBigEndian, LibWebRTCAudioFormat::isNonInterleaved);
- return streamFormat;
-}
-
-void RealtimeIncomingAudioSource::OnData(const void* audioData, int bitsPerSample, int sampleRate, size_t numberOfChannels, size_t numberOfFrames)
-{
- // We may receive OnData calls with empty sound data (mono, samples equal to zero and sampleRate equal to 16000) when starting the call.
- // FIXME: For the moment we skip them, we should find a better solution at libwebrtc level to not be called until getting some real data.
- if (sampleRate == 16000 && numberOfChannels == 1)
- return;
-
- ASSERT(bitsPerSample == 16);
- ASSERT(numberOfChannels == 1 || numberOfChannels == 2);
- ASSERT(sampleRate == 48000);
-
- CMTime startTime = CMTimeMake(m_numberOfFrames, sampleRate);
- auto mediaTime = PAL::toMediaTime(startTime);
- m_numberOfFrames += numberOfFrames;
-
- AudioStreamBasicDescription newDescription = streamDescription(sampleRate, numberOfChannels);
-
- // FIXME: We should not need to do the extra memory allocation and copy.
- // Instead, we should be able to directly pass audioData pointer.
- WebAudioBufferList audioBufferList { CAAudioStreamDescription(newDescription), WTF::safeCast<uint32_t>(numberOfFrames) };
- audioBufferList.buffer(0)->mDataByteSize = numberOfChannels * numberOfFrames * bitsPerSample / 8;
- audioBufferList.buffer(0)->mNumberChannels = numberOfChannels;
-
- if (muted())
- memset(audioBufferList.buffer(0)->mData, 0, audioBufferList.buffer(0)->mDataByteSize);
- else
- memcpy(audioBufferList.buffer(0)->mData, audioData, audioBufferList.buffer(0)->mDataByteSize);
-
- audioSamplesAvailable(mediaTime, audioBufferList, CAAudioStreamDescription(newDescription), numberOfFrames);
-}
-
-void RealtimeIncomingAudioSource::startProducingData()
-{
- if (m_audioTrack)
- m_audioTrack->AddSink(this);
-}
-
-void RealtimeIncomingAudioSource::stopProducingData()
-{
- if (m_audioTrack)
- m_audioTrack->RemoveSink(this);
-}
-
-void RealtimeIncomingAudioSource::setSourceTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& track)
-{
- ASSERT(!m_audioTrack);
- ASSERT(track);
-
- m_audioTrack = WTFMove(track);
- notifyMutedChange(!m_audioTrack);
- if (isProducingData())
- m_audioTrack->AddSink(this);
-}
-
-const RealtimeMediaSourceCapabilities& RealtimeIncomingAudioSource::capabilities() const
-{
- return RealtimeMediaSourceCapabilities::emptyCapabilities();
-}
-
-const RealtimeMediaSourceSettings& RealtimeIncomingAudioSource::settings() const
-{
- return m_currentSettings;
-}
-
-}
-
-#endif // USE(LIBWEBRTC)
Deleted: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h (223411 => 223412)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h 2017-10-16 16:38:15 UTC (rev 223411)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h 2017-10-16 16:40:48 UTC (rev 223412)
@@ -1,75 +0,0 @@
-/*
- * Copyright (C) 2017 Apple Inc. All rights reserved.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions
- * are met:
- *
- * 1. Redistributions of source code must retain the above copyright
- * notice, this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright
- * notice, this list of conditions and the following disclaimer
- * in the documentation and/or other materials provided with the
- * distribution.
- * 3. Neither the name of Ericsson nor the names of its contributors
- * may be used to endorse or promote products derived from this
- * software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
- * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
- * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
- * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
- * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
- * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
- * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
- * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
- * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
- * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#pragma once
-
-#if USE(LIBWEBRTC)
-
-#include "LibWebRTCMacros.h"
-#include "RealtimeMediaSource.h"
-#include <CoreAudio/CoreAudioTypes.h>
-#include <webrtc/api/mediastreaminterface.h>
-#include <wtf/RetainPtr.h>
-
-typedef const struct opaqueCMFormatDescription *CMFormatDescriptionRef;
-
-namespace WebCore {
-
-class WebAudioSourceProviderAVFObjC;
-
-class RealtimeIncomingAudioSource final : public RealtimeMediaSource, private webrtc::AudioTrackSinkInterface {
-public:
- static Ref<RealtimeIncomingAudioSource> create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&&, String&&);
-
- void setSourceTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>&&);
-
-private:
- RealtimeIncomingAudioSource(rtc::scoped_refptr<webrtc::AudioTrackInterface>&&, String&&);
- ~RealtimeIncomingAudioSource();
-
- // webrtc::AudioTrackSinkInterface API
- void OnData(const void* audioData, int bitsPerSample, int sampleRate, size_t numberOfChannels, size_t numberOfFrames) final;
-
- // RealtimeMediaSource API
- void startProducingData() final;
- void stopProducingData() final;
-
- const RealtimeMediaSourceCapabilities& capabilities() const final;
- const RealtimeMediaSourceSettings& settings() const final;
-
- RealtimeMediaSourceSettings m_currentSettings;
- rtc::scoped_refptr<webrtc::AudioTrackInterface> m_audioTrack;
-
- uint64_t m_numberOfFrames { 0 };
-};
-
-} // namespace WebCore
-
-#endif // USE(LIBWEBRTC)
Copied: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.cpp (from rev 223411, trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp) (0 => 223412)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.cpp (rev 0)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.cpp 2017-10-16 16:40:48 UTC (rev 223412)
@@ -0,0 +1,101 @@
+/*
+ * Copyright (C) 2017 Apple Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer
+ * in the documentation and/or other materials provided with the
+ * distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "config.h"
+#include "RealtimeIncomingAudioSourceCocoa.h"
+
+#if USE(LIBWEBRTC)
+
+#include "AudioStreamDescription.h"
+#include "CAAudioStreamDescription.h"
+#include "LibWebRTCAudioFormat.h"
+#include "WebAudioBufferList.h"
+#include "WebAudioSourceProviderAVFObjC.h"
+#include <pal/avfoundation/MediaTimeAVFoundation.h>
+
+#include "CoreMediaSoftLink.h"
+
+namespace WebCore {
+
+Ref<RealtimeIncomingAudioSource> RealtimeIncomingAudioSource::create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
+{
+ auto source = RealtimeIncomingAudioSourceCocoa::create(WTFMove(audioTrack), WTFMove(audioTrackId));
+ source->start();
+ return WTFMove(source);
+}
+
+Ref<RealtimeIncomingAudioSourceCocoa> RealtimeIncomingAudioSourceCocoa::create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
+{
+ return adoptRef(*new RealtimeIncomingAudioSourceCocoa(WTFMove(audioTrack), WTFMove(audioTrackId)));
+}
+
+RealtimeIncomingAudioSourceCocoa::RealtimeIncomingAudioSourceCocoa(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
+ : RealtimeIncomingAudioSource(WTFMove(audioTrack), WTFMove(audioTrackId))
+{
+}
+
+static inline AudioStreamBasicDescription streamDescription(size_t sampleRate, size_t channelCount)
+{
+ AudioStreamBasicDescription streamFormat;
+ FillOutASBDForLPCM(streamFormat, sampleRate, channelCount, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::isFloat, LibWebRTCAudioFormat::isBigEndian, LibWebRTCAudioFormat::isNonInterleaved);
+ return streamFormat;
+}
+
+void RealtimeIncomingAudioSourceCocoa::OnData(const void* audioData, int bitsPerSample, int sampleRate, size_t numberOfChannels, size_t numberOfFrames)
+{
+ // We may receive OnData calls with empty sound data (mono, samples equal to zero and sampleRate equal to 16000) when starting the call.
+ // FIXME: For the moment we skip them, we should find a better solution at libwebrtc level to not be called until getting some real data.
+ if (sampleRate == 16000 && numberOfChannels == 1)
+ return;
+
+ ASSERT(bitsPerSample == 16);
+ ASSERT(numberOfChannels == 1 || numberOfChannels == 2);
+ ASSERT(sampleRate == 48000);
+
+ CMTime startTime = CMTimeMake(m_numberOfFrames, sampleRate);
+ auto mediaTime = PAL::toMediaTime(startTime);
+ m_numberOfFrames += numberOfFrames;
+
+ AudioStreamBasicDescription newDescription = streamDescription(sampleRate, numberOfChannels);
+
+ // FIXME: We should not need to do the extra memory allocation and copy.
+ // Instead, we should be able to directly pass audioData pointer.
+ WebAudioBufferList audioBufferList { CAAudioStreamDescription(newDescription), WTF::safeCast<uint32_t>(numberOfFrames) };
+ audioBufferList.buffer(0)->mDataByteSize = numberOfChannels * numberOfFrames * bitsPerSample / 8;
+ audioBufferList.buffer(0)->mNumberChannels = numberOfChannels;
+
+ if (muted())
+ memset(audioBufferList.buffer(0)->mData, 0, audioBufferList.buffer(0)->mDataByteSize);
+ else
+ memcpy(audioBufferList.buffer(0)->mData, audioData, audioBufferList.buffer(0)->mDataByteSize);
+
+ audioSamplesAvailable(mediaTime, audioBufferList, CAAudioStreamDescription(newDescription), numberOfFrames);
+}
+
+}
+
+#endif // USE(LIBWEBRTC)
Copied: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.h (from rev 223411, trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.h) (0 => 223412)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.h (rev 0)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.h 2017-10-16 16:40:48 UTC (rev 223412)
@@ -0,0 +1,57 @@
+/*
+ * Copyright (C) 2017 Apple Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer
+ * in the documentation and/or other materials provided with the
+ * distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#pragma once
+
+#if USE(LIBWEBRTC)
+
+#include "RealtimeIncomingAudioSource.h"
+
+#include <CoreAudio/CoreAudioTypes.h>
+
+typedef const struct opaqueCMFormatDescription *CMFormatDescriptionRef;
+
+namespace WebCore {
+
+class WebAudioSourceProviderAVFObjC;
+
+class RealtimeIncomingAudioSourceCocoa final : public RealtimeIncomingAudioSource {
+public:
+ static Ref<RealtimeIncomingAudioSourceCocoa> create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&&, String&&);
+
+private:
+ RealtimeIncomingAudioSourceCocoa(rtc::scoped_refptr<webrtc::AudioTrackInterface>&&, String&&);
+
+ // webrtc::AudioTrackSinkInterface API
+ void OnData(const void* audioData, int bitsPerSample, int sampleRate, size_t numberOfChannels, size_t numberOfFrames) final;
+
+ uint64_t m_numberOfFrames { 0 };
+};
+
+} // namespace WebCore
+
+#endif // USE(LIBWEBRTC)
Deleted: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp (223411 => 223412)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-10-16 16:38:15 UTC (rev 223411)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-10-16 16:40:48 UTC (rev 223412)
@@ -1,203 +0,0 @@
-/*
- * Copyright (C) 2017 Apple Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted, provided that the following conditions
- * are required to be met:
- *
- * 1. Redistributions of source code must retain the above copyright
- * notice, this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright
- * notice, this list of conditions and the following disclaimer in the
- * documentation and/or other materials provided with the distribution.
- * 3. Neither the name of Apple Inc. nor the names of
- * its contributors may be used to endorse or promote products derived
- * from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY
- * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
- * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
- * DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR
- * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
- * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
- * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
- * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
- * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
- * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "config.h"
-#include "RealtimeOutgoingAudioSource.h"
-
-#if USE(LIBWEBRTC)
-
-#include "CAAudioStreamDescription.h"
-#include "LibWebRTCAudioFormat.h"
-#include "LibWebRTCProvider.h"
-
-namespace WebCore {
-
-static inline AudioStreamBasicDescription libwebrtcAudioFormat(Float64 sampleRate, size_t channelCount)
-{
- // FIXME: Microphones can have more than two channels. In such case, we should do the mix down based on AudioChannelLayoutTag.
- size_t libWebRTCChannelCount = channelCount >= 2 ? 2 : channelCount;
- AudioStreamBasicDescription streamFormat;
- FillOutASBDForLPCM(streamFormat, sampleRate, libWebRTCChannelCount, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::isFloat, LibWebRTCAudioFormat::isBigEndian, LibWebRTCAudioFormat::isNonInterleaved);
- return streamFormat;
-}
-
-RealtimeOutgoingAudioSource::RealtimeOutgoingAudioSource(Ref<MediaStreamTrackPrivate>&& audioSource)
- : m_audioSource(WTFMove(audioSource))
- , m_sampleConverter(AudioSampleDataSource::create(LibWebRTCAudioFormat::sampleRate * 2))
- , m_silenceAudioTimer(*this, &RealtimeOutgoingAudioSource::sendSilence)
-{
- m_audioSource->addObserver(*this);
- initializeConverter();
-}
-
-bool RealtimeOutgoingAudioSource::setSource(Ref<MediaStreamTrackPrivate>&& newSource)
-{
- m_audioSource->removeObserver(*this);
- m_audioSource = WTFMove(newSource);
- m_audioSource->addObserver(*this);
-
- initializeConverter();
- return true;
-}
-
-void RealtimeOutgoingAudioSource::initializeConverter()
-{
- m_muted = m_audioSource->muted();
- m_enabled = m_audioSource->enabled();
- handleMutedIfNeeded();
-}
-
-void RealtimeOutgoingAudioSource::stop()
-{
- m_silenceAudioTimer.stop();
- m_audioSource->removeObserver(*this);
-}
-
-void RealtimeOutgoingAudioSource::sourceMutedChanged()
-{
- m_muted = m_audioSource->muted();
- handleMutedIfNeeded();
-}
-
-void RealtimeOutgoingAudioSource::sourceEnabledChanged()
-{
- m_enabled = m_audioSource->enabled();
- handleMutedIfNeeded();
-}
-
-void RealtimeOutgoingAudioSource::handleMutedIfNeeded()
-{
- bool isSilenced = m_muted || !m_enabled;
- m_sampleConverter->setMuted(isSilenced);
- if (isSilenced && !m_silenceAudioTimer.isActive())
- m_silenceAudioTimer.startRepeating(1_s);
- if (!isSilenced && m_silenceAudioTimer.isActive())
- m_silenceAudioTimer.stop();
-}
-
-void RealtimeOutgoingAudioSource::sendSilence()
-{
- LibWebRTCProvider::callOnWebRTCSignalingThread([this, protectedThis = makeRef(*this)] {
- size_t chunkSampleCount = m_outputStreamDescription.sampleRate() / 100;
- size_t bufferSize = chunkSampleCount * LibWebRTCAudioFormat::sampleByteSize * m_outputStreamDescription.numberOfChannels();
-
- if (!bufferSize)
- return;
-
- m_audioBuffer.grow(bufferSize);
- memset(m_audioBuffer.data(), 0, bufferSize);
- for (auto sink : m_sinks)
- sink->OnData(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, m_outputStreamDescription.sampleRate(), m_outputStreamDescription.numberOfChannels(), chunkSampleCount);
- });
-}
-
-bool RealtimeOutgoingAudioSource::isReachingBufferedAudioDataHighLimit()
-{
- auto writtenAudioDuration = m_writeCount / m_inputStreamDescription.sampleRate();
- auto readAudioDuration = m_readCount / m_outputStreamDescription.sampleRate();
-
- ASSERT(writtenAudioDuration >= readAudioDuration);
- return writtenAudioDuration > readAudioDuration + 0.5;
-}
-
-bool RealtimeOutgoingAudioSource::isReachingBufferedAudioDataLowLimit()
-{
- auto writtenAudioDuration = m_writeCount / m_inputStreamDescription.sampleRate();
- auto readAudioDuration = m_readCount / m_outputStreamDescription.sampleRate();
-
- ASSERT(writtenAudioDuration >= readAudioDuration);
- return writtenAudioDuration < readAudioDuration + 0.1;
-}
-
-bool RealtimeOutgoingAudioSource::hasBufferedEngouhData()
-{
- auto writtenAudioDuration = m_writeCount / m_inputStreamDescription.sampleRate();
- auto readAudioDuration = m_readCount / m_outputStreamDescription.sampleRate();
-
- ASSERT(writtenAudioDuration >= readAudioDuration);
- return writtenAudioDuration >= readAudioDuration + 0.01;
-}
-
-void RealtimeOutgoingAudioSource::audioSamplesAvailable(const MediaTime&, const PlatformAudioData& audioData, const AudioStreamDescription& streamDescription, size_t sampleCount)
-{
- if (m_inputStreamDescription != streamDescription) {
- m_inputStreamDescription = toCAAudioStreamDescription(streamDescription);
- auto status = m_sampleConverter->setInputFormat(m_inputStreamDescription);
- ASSERT_UNUSED(status, !status);
-
- m_outputStreamDescription = libwebrtcAudioFormat(LibWebRTCAudioFormat::sampleRate, streamDescription.numberOfChannels());
- status = m_sampleConverter->setOutputFormat(m_outputStreamDescription.streamDescription());
- ASSERT(!status);
- }
-
- // Let's skip pushing samples if we are too slow pulling them.
- if (m_skippingAudioData) {
- if (!isReachingBufferedAudioDataLowLimit())
- return;
- m_skippingAudioData = false;
- } else if (isReachingBufferedAudioDataHighLimit()) {
- m_skippingAudioData = true;
- return;
- }
-
- // If we change the audio track or its sample rate changes, the timestamp based on m_writeCount may be wrong.
- // FIXME: We should update m_writeCount to be valid according the new sampleRate.
- m_sampleConverter->pushSamples(MediaTime(m_writeCount, static_cast<uint32_t>(m_inputStreamDescription.sampleRate())), audioData, sampleCount);
- m_writeCount += sampleCount;
-
- if (!hasBufferedEngouhData())
- return;
-
- LibWebRTCProvider::callOnWebRTCSignalingThread([protectedThis = makeRef(*this)] {
- protectedThis->pullAudioData();
- });
-}
-
-void RealtimeOutgoingAudioSource::pullAudioData()
-{
- // libwebrtc expects 10 ms chunks.
- size_t chunkSampleCount = m_outputStreamDescription.sampleRate() / 100;
- size_t bufferSize = chunkSampleCount * LibWebRTCAudioFormat::sampleByteSize * m_outputStreamDescription.numberOfChannels();
- m_audioBuffer.grow(bufferSize);
-
- AudioBufferList bufferList;
- bufferList.mNumberBuffers = 1;
- bufferList.mBuffers[0].mNumberChannels = m_outputStreamDescription.numberOfChannels();
- bufferList.mBuffers[0].mDataByteSize = bufferSize;
- bufferList.mBuffers[0].mData = m_audioBuffer.data();
-
- m_sampleConverter->pullAvalaibleSamplesAsChunks(bufferList, chunkSampleCount, m_readCount, [this, chunkSampleCount] {
- m_readCount += chunkSampleCount;
- for (auto sink : m_sinks)
- sink->OnData(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, m_outputStreamDescription.sampleRate(), m_outputStreamDescription.numberOfChannels(), chunkSampleCount);
- });
-}
-
-} // namespace WebCore
-
-#endif // USE(LIBWEBRTC)
Deleted: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h (223411 => 223412)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h 2017-10-16 16:38:15 UTC (rev 223411)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h 2017-10-16 16:40:48 UTC (rev 223412)
@@ -1,108 +0,0 @@
-/*
- * Copyright (C) 2017 Apple Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted, provided that the following conditions
- * are required to be met:
- *
- * 1. Redistributions of source code must retain the above copyright
- * notice, this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright
- * notice, this list of conditions and the following disclaimer in the
- * documentation and/or other materials provided with the distribution.
- * 3. Neither the name of Apple Inc. nor the names of
- * its contributors may be used to endorse or promote products derived
- * from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY
- * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
- * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
- * DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR
- * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
- * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
- * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
- * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
- * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
- * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#pragma once
-
-#if USE(LIBWEBRTC)
-
-#include "AudioSampleDataSource.h"
-#include "LibWebRTCMacros.h"
-#include "MediaStreamTrackPrivate.h"
-#include "Timer.h"
-#include <webrtc/api/mediastreaminterface.h>
-#include <wtf/ThreadSafeRefCounted.h>
-
-namespace webrtc {
-class AudioTrackInterface;
-class AudioTrackSinkInterface;
-}
-
-namespace WebCore {
-
-class RealtimeOutgoingAudioSource final : public ThreadSafeRefCounted<RealtimeOutgoingAudioSource>, public webrtc::AudioSourceInterface, private MediaStreamTrackPrivate::Observer {
-public:
- static Ref<RealtimeOutgoingAudioSource> create(Ref<MediaStreamTrackPrivate>&& audioSource) { return adoptRef(*new RealtimeOutgoingAudioSource(WTFMove(audioSource))); }
- ~RealtimeOutgoingAudioSource() { stop(); }
-
- void stop();
-
- bool setSource(Ref<MediaStreamTrackPrivate>&&);
- MediaStreamTrackPrivate& source() const { return m_audioSource.get(); }
-
-private:
- explicit RealtimeOutgoingAudioSource(Ref<MediaStreamTrackPrivate>&&);
-
- virtual void AddSink(webrtc::AudioTrackSinkInterface* sink) { m_sinks.append(sink); }
- virtual void RemoveSink(webrtc::AudioTrackSinkInterface* sink) { m_sinks.removeFirst(sink); }
-
- int AddRef() const final { ref(); return refCount(); }
- int Release() const final { deref(); return refCount(); }
- SourceState state() const final { return kLive; }
- bool remote() const final { return false; }
- void RegisterObserver(webrtc::ObserverInterface*) final { }
- void UnregisterObserver(webrtc::ObserverInterface*) final { }
-
- void sourceMutedChanged();
- void sourceEnabledChanged();
- void audioSamplesAvailable(const MediaTime&, const PlatformAudioData&, const AudioStreamDescription&, size_t);
-
- bool isReachingBufferedAudioDataHighLimit();
- bool isReachingBufferedAudioDataLowLimit();
- bool hasBufferedEngouhData();
-
- // MediaStreamTrackPrivate::Observer API
- void trackMutedChanged(MediaStreamTrackPrivate&) final { sourceMutedChanged(); }
- void trackEnabledChanged(MediaStreamTrackPrivate&) final { sourceEnabledChanged(); }
- void audioSamplesAvailable(MediaStreamTrackPrivate&, const MediaTime& mediaTime, const PlatformAudioData& data, const AudioStreamDescription& description, size_t sampleCount) final { audioSamplesAvailable(mediaTime, data, description, sampleCount); }
- void trackEnded(MediaStreamTrackPrivate&) final { }
- void trackSettingsChanged(MediaStreamTrackPrivate&) final { }
-
- void pullAudioData();
-
- void initializeConverter();
- void handleMutedIfNeeded();
- void sendSilence();
-
- Vector<webrtc::AudioTrackSinkInterface*> m_sinks;
- Ref<MediaStreamTrackPrivate> m_audioSource;
- Ref<AudioSampleDataSource> m_sampleConverter;
- CAAudioStreamDescription m_inputStreamDescription;
- CAAudioStreamDescription m_outputStreamDescription;
-
- Vector<uint8_t> m_audioBuffer;
- uint64_t m_readCount { 0 };
- uint64_t m_writeCount { 0 };
- bool m_muted { false };
- bool m_enabled { true };
- bool m_skippingAudioData { false };
- Timer m_silenceAudioTimer;
-};
-
-} // namespace WebCore
-
-#endif // USE(LIBWEBRTC)
Copied: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSourceCocoa.cpp (from rev 223411, trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp) (0 => 223412)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSourceCocoa.cpp (rev 0)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSourceCocoa.cpp 2017-10-16 16:40:48 UTC (rev 223412)
@@ -0,0 +1,165 @@
+/*
+ * Copyright (C) 2017 Apple Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted, provided that the following conditions
+ * are required to be met:
+ *
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+ * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "config.h"
+#include "RealtimeOutgoingAudioSourceCocoa.h"
+
+#if USE(LIBWEBRTC)
+
+#include "CAAudioStreamDescription.h"
+#include "LibWebRTCAudioFormat.h"
+#include "LibWebRTCProvider.h"
+
+namespace WebCore {
+
+static inline AudioStreamBasicDescription libwebrtcAudioFormat(Float64 sampleRate, size_t channelCount)
+{
+ // FIXME: Microphones can have more than two channels. In such case, we should do the mix down based on AudioChannelLayoutTag.
+ size_t libWebRTCChannelCount = channelCount >= 2 ? 2 : channelCount;
+ AudioStreamBasicDescription streamFormat;
+ FillOutASBDForLPCM(streamFormat, sampleRate, libWebRTCChannelCount, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::isFloat, LibWebRTCAudioFormat::isBigEndian, LibWebRTCAudioFormat::isNonInterleaved);
+ return streamFormat;
+}
+
+RealtimeOutgoingAudioSourceCocoa::RealtimeOutgoingAudioSourceCocoa(Ref<MediaStreamTrackPrivate>&& audioSource)
+ : RealtimeOutgoingAudioSource(WTFMove(audioSource))
+ , m_sampleConverter(AudioSampleDataSource::create(LibWebRTCAudioFormat::sampleRate * 2))
+{
+}
+
+Ref<RealtimeOutgoingAudioSource> RealtimeOutgoingAudioSource::create(Ref<MediaStreamTrackPrivate>&& audioSource)
+{
+ return RealtimeOutgoingAudioSourceCocoa::create(WTFMove(audioSource));
+}
+
+void RealtimeOutgoingAudioSourceCocoa::handleMutedIfNeeded()
+{
+ bool isSilenced = m_muted || !m_enabled;
+ m_sampleConverter->setMuted(isSilenced);
+
+ RealtimeOutgoingAudioSource::handleMutedIfNeeded();
+}
+
+void RealtimeOutgoingAudioSourceCocoa::sendSilence()
+{
+ LibWebRTCProvider::callOnWebRTCSignalingThread([this, protectedThis = makeRef(*this)] {
+ size_t chunkSampleCount = m_outputStreamDescription.sampleRate() / 100;
+ size_t bufferSize = chunkSampleCount * LibWebRTCAudioFormat::sampleByteSize * m_outputStreamDescription.numberOfChannels();
+
+ if (!bufferSize)
+ return;
+
+ m_audioBuffer.grow(bufferSize);
+ memset(m_audioBuffer.data(), 0, bufferSize);
+ for (auto sink : m_sinks)
+ sink->OnData(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, m_outputStreamDescription.sampleRate(), m_outputStreamDescription.numberOfChannels(), chunkSampleCount);
+ });
+}
+
+bool RealtimeOutgoingAudioSourceCocoa::isReachingBufferedAudioDataHighLimit()
+{
+ auto writtenAudioDuration = m_writeCount / m_inputStreamDescription.sampleRate();
+ auto readAudioDuration = m_readCount / m_outputStreamDescription.sampleRate();
+
+ ASSERT(writtenAudioDuration >= readAudioDuration);
+ return writtenAudioDuration > readAudioDuration + 0.5;
+}
+
+bool RealtimeOutgoingAudioSourceCocoa::isReachingBufferedAudioDataLowLimit()
+{
+ auto writtenAudioDuration = m_writeCount / m_inputStreamDescription.sampleRate();
+ auto readAudioDuration = m_readCount / m_outputStreamDescription.sampleRate();
+
+ ASSERT(writtenAudioDuration >= readAudioDuration);
+ return writtenAudioDuration < readAudioDuration + 0.1;
+}
+
+bool RealtimeOutgoingAudioSourceCocoa::hasBufferedEnoughData()
+{
+ auto writtenAudioDuration = m_writeCount / m_inputStreamDescription.sampleRate();
+ auto readAudioDuration = m_readCount / m_outputStreamDescription.sampleRate();
+
+ ASSERT(writtenAudioDuration >= readAudioDuration);
+ return writtenAudioDuration >= readAudioDuration + 0.01;
+}
+
+void RealtimeOutgoingAudioSourceCocoa::audioSamplesAvailable(const MediaTime&, const PlatformAudioData& audioData, const AudioStreamDescription& streamDescription, size_t sampleCount)
+{
+ if (m_inputStreamDescription != streamDescription) {
+ m_inputStreamDescription = toCAAudioStreamDescription(streamDescription);
+ auto status = m_sampleConverter->setInputFormat(m_inputStreamDescription);
+ ASSERT_UNUSED(status, !status);
+
+ m_outputStreamDescription = libwebrtcAudioFormat(LibWebRTCAudioFormat::sampleRate, streamDescription.numberOfChannels());
+ status = m_sampleConverter->setOutputFormat(m_outputStreamDescription.streamDescription());
+ ASSERT(!status);
+ }
+
+ // Let's skip pushing samples if we are too slow pulling them.
+ if (m_skippingAudioData) {
+ if (!isReachingBufferedAudioDataLowLimit())
+ return;
+ m_skippingAudioData = false;
+ } else if (isReachingBufferedAudioDataHighLimit()) {
+ m_skippingAudioData = true;
+ return;
+ }
+
+ // If we change the audio track or its sample rate changes, the timestamp based on m_writeCount may be wrong.
+ // FIXME: We should update m_writeCount to be valid according the new sampleRate.
+ m_sampleConverter->pushSamples(MediaTime(m_writeCount, static_cast<uint32_t>(m_inputStreamDescription.sampleRate())), audioData, sampleCount);
+ m_writeCount += sampleCount;
+
+ if (!hasBufferedEnoughData())
+ return;
+
+ LibWebRTCProvider::callOnWebRTCSignalingThread([protectedThis = makeRef(*this)] {
+ protectedThis->pullAudioData();
+ });
+}
+
+void RealtimeOutgoingAudioSourceCocoa::pullAudioData()
+{
+ // libwebrtc expects 10 ms chunks.
+ size_t chunkSampleCount = m_outputStreamDescription.sampleRate() / 100;
+ size_t bufferSize = chunkSampleCount * LibWebRTCAudioFormat::sampleByteSize * m_outputStreamDescription.numberOfChannels();
+ m_audioBuffer.grow(bufferSize);
+
+ AudioBufferList bufferList;
+ bufferList.mNumberBuffers = 1;
+ bufferList.mBuffers[0].mNumberChannels = m_outputStreamDescription.numberOfChannels();
+ bufferList.mBuffers[0].mDataByteSize = bufferSize;
+ bufferList.mBuffers[0].mData = m_audioBuffer.data();
+
+ m_sampleConverter->pullAvalaibleSamplesAsChunks(bufferList, chunkSampleCount, m_readCount, [this, chunkSampleCount] {
+ m_readCount += chunkSampleCount;
+ for (auto sink : m_sinks)
+ sink->OnData(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, m_outputStreamDescription.sampleRate(), m_outputStreamDescription.numberOfChannels(), chunkSampleCount);
+ });
+}
+
+} // namespace WebCore
+
+#endif // USE(LIBWEBRTC)
Added: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSourceCocoa.h (0 => 223412)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSourceCocoa.h (rev 0)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSourceCocoa.h 2017-10-16 16:40:48 UTC (rev 223412)
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2017 Apple Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted, provided that the following conditions
+ * are required to be met:
+ *
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+ * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#pragma once
+
+#if USE(LIBWEBRTC)
+
+#include "AudioSampleDataSource.h"
+#include "RealtimeOutgoingAudioSource.h"
+
+namespace webrtc {
+class AudioTrackInterface;
+class AudioTrackSinkInterface;
+}
+
+namespace WebCore {
+
+class RealtimeOutgoingAudioSourceCocoa final : public RealtimeOutgoingAudioSource {
+public:
+ static Ref<RealtimeOutgoingAudioSourceCocoa> create(Ref<MediaStreamTrackPrivate>&& audioSource) { return adoptRef(*new RealtimeOutgoingAudioSourceCocoa(WTFMove(audioSource))); }
+
+private:
+ explicit RealtimeOutgoingAudioSourceCocoa(Ref<MediaStreamTrackPrivate>&&);
+
+ void audioSamplesAvailable(const MediaTime&, const PlatformAudioData&, const AudioStreamDescription&, size_t) final;
+
+ bool isReachingBufferedAudioDataHighLimit() final;
+ bool isReachingBufferedAudioDataLowLimit() final;
+ bool hasBufferedEnoughData() final;
+
+ void pullAudioData() final;
+
+ void handleMutedIfNeeded() final;
+ void sendSilence() final;
+
+ Ref<AudioSampleDataSource> m_sampleConverter;
+ CAAudioStreamDescription m_inputStreamDescription;
+ CAAudioStreamDescription m_outputStreamDescription;
+
+ Vector<uint8_t> m_audioBuffer;
+ uint64_t m_readCount { 0 };
+ uint64_t m_writeCount { 0 };
+ bool m_skippingAudioData { false };
+};
+
+} // namespace WebCore
+
+#endif // USE(LIBWEBRTC)
