Diff
Modified: trunk/Source/WebCore/ChangeLog (211958 => 211959)
--- trunk/Source/WebCore/ChangeLog 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/ChangeLog 2017-02-09 16:33:06 UTC (rev 211959)
@@ -1,3 +1,69 @@
+2017-02-09 Jer Noble <jer.no...@apple.com>
+
+ Configure MockRealtimeAudioSourceMac to generate stereo audio
+ https://bugs.webkit.org/show_bug.cgi?id=168027
+
+ Reviewed by Eric Carlson.
+
+ Update MockRealtimeAudioSourceMac to generate stereo audio.
+
+ First, because the pattern of creating a AudioBufferList structure (with all it's quirks and
+ weird requirements) was repeated multiple places, add a new wrapper around ABL called
+ WebAudioBufferList which takes care of correctly initializing the ABL structure and manages
+ the lifetime of its data members.
+
+ * WebCore.xcodeproj/project.pbxproj:
+ * platform/audio/PlatformAudioData.h: Added.
+ (WebCore::PlatformAudioData::kind):
+ * platform/audio/WebAudioBufferList.cpp: Added.
+ (WebCore::WebAudioBufferList::WebAudioBufferList):
+ (WebCore::WebAudioBufferList::buffers):
+ (WebCore::WebAudioBufferList::bufferCount):
+ (WebCore::WebAudioBufferList::buffer):
+ * platform/audio/WebAudioBufferList.h: Added.
+ (WebCore::WebAudioBufferList::list):
+ (WebCore::WebAudioBufferList::operator AudioBufferList&):
+ (WebCore::WebAudioBufferList::kind):
+ (isType):
+
+ Then update existing code to work in terms of WebAudioBufferList:
+
+ * platform/audio/mac/AudioSampleBufferList.cpp:
+ (WebCore::AudioSampleBufferList::AudioSampleBufferList):
+ (WebCore::AudioSampleBufferList::mixFrom):
+ (WebCore::AudioSampleBufferList::copyFrom):
+ (WebCore::AudioSampleBufferList::copyTo):
+ (WebCore::AudioSampleBufferList::reset):
+ (WebCore::AudioSampleBufferList::configureBufferListForStream): Deleted.
+ * platform/audio/mac/AudioSampleBufferList.h:
+ (WebCore::AudioSampleBufferList::bufferList):
+ * platform/audio/mac/AudioSampleDataSource.cpp:
+ (WebCore::AudioSampleDataSource::pushSamples):
+ * platform/audio/mac/AudioSampleDataSource.h:
+ * platform/mediastream/RealtimeMediaSource.cpp:
+ (WebCore::RealtimeMediaSource::audioSamplesAvailable):
+ * platform/mediastream/RealtimeMediaSource.h:
+ (WebCore::RealtimeMediaSource::Observer::audioSamplesAvailable):
+ * platform/mediastream/mac/AVAudioCaptureSource.h:
+ * platform/mediastream/mac/AVAudioCaptureSource.mm:
+ (WebCore::AVAudioCaptureSource::captureOutputDidOutputSampleBufferFromConnection):
+ * platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.cpp:
+ (WebCore::AudioTrackPrivateMediaStreamCocoa::audioSamplesAvailable):
+ * platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.h:
+ * platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp:
+ (WebCore::RealtimeOutgoingAudioSource::audioSamplesAvailable):
+ * platform/mediastream/mac/RealtimeOutgoingAudioSource.h:
+
+ Finally, actually update MockRealtimeAudioSource to emit stereo samples. Importantly, set
+ the correct values for the m_streamFormat; mBytesPerFrame and mBytesPerPacket are not
+ multiplied by the channelCount. When generating audio, write to both channels of data.
+
+ * platform/mediastream/mac/MockRealtimeAudioSourceMac.h:
+ * platform/mediastream/mac/MockRealtimeAudioSourceMac.mm:
+ (WebCore::MockRealtimeAudioSourceMac::emitSampleBuffers):
+ (WebCore::MockRealtimeAudioSourceMac::reconfigure):
+ (WebCore::MockRealtimeAudioSourceMac::render):
+
2017-02-09 Antti Koivisto <an...@apple.com>
Nullptr crash under styleForFirstLetter
Modified: trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj (211958 => 211959)
--- trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2017-02-09 16:33:06 UTC (rev 211959)
@@ -5971,6 +5971,8 @@
CDE595951BF16DF300A1CBE8 /* CDMSessionAVContentKeySession.mm in Sources */ = {isa = PBXBuildFile; fileRef = CDE595931BF166AD00A1CBE8 /* CDMSessionAVContentKeySession.mm */; };
CDE595971BF26E2100A1CBE8 /* CDMSessionMediaSourceAVFObjC.h in Headers */ = {isa = PBXBuildFile; fileRef = CDE595961BF26E2100A1CBE8 /* CDMSessionMediaSourceAVFObjC.h */; };
CDE5959D1BF2757100A1CBE8 /* CDMSessionMediaSourceAVFObjC.mm in Sources */ = {isa = PBXBuildFile; fileRef = CDE5959C1BF2757100A1CBE8 /* CDMSessionMediaSourceAVFObjC.mm */; };
+ CDE667A41E4BBF1500E8154A /* WebAudioBufferList.cpp in Sources */ = {isa = PBXBuildFile; fileRef = CDE667A21E4BBF1500E8154A /* WebAudioBufferList.cpp */; };
+ CDE667A51E4BBF1500E8154A /* WebAudioBufferList.h in Headers */ = {isa = PBXBuildFile; fileRef = CDE667A31E4BBF1500E8154A /* WebAudioBufferList.h */; };
CDE7FC44181904B1002BBB77 /* OrderIterator.cpp in Sources */ = {isa = PBXBuildFile; fileRef = CDE7FC42181904B1002BBB77 /* OrderIterator.cpp */; };
CDE7FC45181904B1002BBB77 /* OrderIterator.h in Headers */ = {isa = PBXBuildFile; fileRef = CDE7FC43181904B1002BBB77 /* OrderIterator.h */; settings = {ATTRIBUTES = (Private, ); }; };
CDE83DB1183C44060031EAA3 /* VideoPlaybackQuality.cpp in Sources */ = {isa = PBXBuildFile; fileRef = CDE83DAF183C44060031EAA3 /* VideoPlaybackQuality.cpp */; };
@@ -14087,6 +14089,9 @@
CDE595961BF26E2100A1CBE8 /* CDMSessionMediaSourceAVFObjC.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CDMSessionMediaSourceAVFObjC.h; sourceTree = "<group>"; };
CDE5959C1BF2757100A1CBE8 /* CDMSessionMediaSourceAVFObjC.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = CDMSessionMediaSourceAVFObjC.mm; sourceTree = "<group>"; };
CDE6560E17CA6E7600526BA7 /* mediaControlsApple.js */ = {isa = PBXFileReference; lastKnownFileType = sourcecode._javascript_; path = mediaControlsApple.js; sourceTree = "<group>"; };
+ CDE667A11E4BBA4D00E8154A /* PlatformAudioData.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = PlatformAudioData.h; sourceTree = "<group>"; };
+ CDE667A21E4BBF1500E8154A /* WebAudioBufferList.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = WebAudioBufferList.cpp; sourceTree = "<group>"; };
+ CDE667A31E4BBF1500E8154A /* WebAudioBufferList.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = WebAudioBufferList.h; sourceTree = "<group>"; };
CDE7FC42181904B1002BBB77 /* OrderIterator.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = OrderIterator.cpp; sourceTree = "<group>"; };
CDE7FC43181904B1002BBB77 /* OrderIterator.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = OrderIterator.h; sourceTree = "<group>"; };
CDE83DAF183C44060031EAA3 /* VideoPlaybackQuality.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = VideoPlaybackQuality.cpp; sourceTree = "<group>"; };
@@ -23140,6 +23145,8 @@
isa = PBXGroup;
children = (
CD669D661D232DFF004D1866 /* MediaSessionManagerCocoa.cpp */,
+ CDE667A21E4BBF1500E8154A /* WebAudioBufferList.cpp */,
+ CDE667A31E4BBF1500E8154A /* WebAudioBufferList.h */,
);
name = cocoa;
sourceTree = "<group>";
@@ -25068,6 +25075,7 @@
FDB1700414A2BAB200A2B5D9 /* MultiChannelResampler.h */,
FD31606C12B026F700C1A359 /* Panner.cpp */,
FD31606D12B026F700C1A359 /* Panner.h */,
+ CDE667A11E4BBA4D00E8154A /* PlatformAudioData.h */,
070E091A1875EF71003A1D3C /* PlatformMediaSession.cpp */,
070E09181875ED93003A1D3C /* PlatformMediaSession.h */,
CDAE8C071746B95700532D78 /* PlatformMediaSessionManager.cpp */,
@@ -27069,6 +27077,7 @@
E17B492116A9B8FF001C8839 /* JSTransitionEvent.h in Headers */,
1A750D5D0A90DEE1000FF215 /* JSTreeWalker.h in Headers */,
A86629CF09DA2B47009633A5 /* JSUIEvent.h in Headers */,
+ CDE667A51E4BBF1500E8154A /* WebAudioBufferList.h in Headers */,
465307D01DB6EE4800E4137C /* JSUIEventInit.h in Headers */,
7C73FB12191EF6F4007DE061 /* JSUserMessageHandler.h in Headers */,
7C73FB0D191EF5A8007DE061 /* JSUserMessageHandlersNamespace.h in Headers */,
@@ -30427,6 +30436,7 @@
BC6932730D7E293900AE44D1 /* JSDOMWindowBase.cpp in Sources */,
BCD9C2620C17AA67005C90A2 /* JSDOMWindowCustom.cpp in Sources */,
460CBF351D4BCD0E0092E88E /* JSDOMWindowProperties.cpp in Sources */,
+ CDE667A41E4BBF1500E8154A /* WebAudioBufferList.cpp in Sources */,
BCBFB53C0DCD29CF0019B3E5 /* JSDOMWindowShell.cpp in Sources */,
A1CC11641E493D0100EFA69C /* FileSystemMac.mm in Sources */,
4170A2EA1D8C0CCA00318452 /* JSDOMWrapper.cpp in Sources */,
Added: trunk/Source/WebCore/platform/audio/PlatformAudioData.h (0 => 211959)
--- trunk/Source/WebCore/platform/audio/PlatformAudioData.h (rev 0)
+++ trunk/Source/WebCore/platform/audio/PlatformAudioData.h 2017-02-09 16:33:06 UTC (rev 211959)
@@ -0,0 +1,45 @@
+/*
+ * 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 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. OR 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
+
+namespace WebCore {
+
+class PlatformAudioData {
+public:
+ virtual ~PlatformAudioData() = default;
+
+ enum class Kind {
+ None,
+ WebAudioBufferList,
+ };
+
+ virtual Kind kind() const { return Kind::None; }
+
+protected:
+ PlatformAudioData() = default;
+};
+
+}
Added: trunk/Source/WebCore/platform/audio/WebAudioBufferList.cpp (0 => 211959)
--- trunk/Source/WebCore/platform/audio/WebAudioBufferList.cpp (rev 0)
+++ trunk/Source/WebCore/platform/audio/WebAudioBufferList.cpp 2017-02-09 16:33:06 UTC (rev 211959)
@@ -0,0 +1,100 @@
+/*
+ * 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 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. OR 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 "WebAudioBufferList.h"
+
+#include "CAAudioStreamDescription.h"
+#include "CoreMediaSoftLink.h"
+
+namespace WebCore {
+
+WebAudioBufferList::WebAudioBufferList(const CAAudioStreamDescription& format)
+{
+ // AudioBufferList is a variable-length struct, so create on the heap with a generic new() operator
+ // with a custom size, and initialize the struct manually.
+ uint32_t bufferCount = format.numberOfChannelStreams();
+ uint32_t channelCount = format.numberOfInterleavedChannels();
+
+ uint64_t bufferListSize = offsetof(AudioBufferList, mBuffers) + (sizeof(AudioBuffer) * std::max(1U, bufferCount));
+ ASSERT(bufferListSize <= SIZE_MAX);
+
+ m_listBufferSize = static_cast<size_t>(bufferListSize);
+ m_list = std::unique_ptr<AudioBufferList>(static_cast<AudioBufferList*>(::operator new (m_listBufferSize)));
+ memset(m_list.get(), 0, m_listBufferSize);
+ m_list->mNumberBuffers = bufferCount;
+ for (uint32_t buffer = 0; buffer < bufferCount; ++buffer)
+ m_list->mBuffers[buffer].mNumberChannels = channelCount;
+}
+
+WebAudioBufferList::WebAudioBufferList(const CAAudioStreamDescription& format, uint32_t sampleCount)
+ : WebAudioBufferList(format)
+{
+ if (!sampleCount)
+ return;
+
+ uint32_t bufferCount = format.numberOfChannelStreams();
+ uint32_t channelCount = format.numberOfInterleavedChannels();
+
+ size_t bytesPerBuffer = sampleCount * channelCount * format.bytesPerFrame();
+ m_flatBuffer.reserveInitialCapacity(bufferCount * bytesPerBuffer);
+ auto data = ""
+
+ for (uint32_t buffer = 0; buffer < m_list->mNumberBuffers; ++buffer) {
+ m_list->mBuffers[buffer].mData = data;
+ data += bytesPerBuffer;
+ }
+}
+
+WebAudioBufferList::WebAudioBufferList(const CAAudioStreamDescription& format, CMSampleBufferRef sampleBuffer)
+ : WebAudioBufferList(format)
+{
+ if (!sampleBuffer)
+ return;
+
+ CMBlockBufferRef buffer = nullptr;
+ if (noErr == CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampleBuffer, nullptr, m_list.get(), m_listBufferSize, kCFAllocatorSystemDefault, kCFAllocatorSystemDefault, kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, &buffer))
+ m_blockBuffer = adoptCF(buffer);
+}
+
+WTF::IteratorRange<AudioBuffer*> WebAudioBufferList::buffers() const
+{
+ return WTF::makeIteratorRange(&m_list->mBuffers[0], &m_list->mBuffers[m_list->mNumberBuffers]);
+}
+
+uint32_t WebAudioBufferList::bufferCount() const
+{
+ return m_list->mNumberBuffers;
+}
+
+AudioBuffer* WebAudioBufferList::buffer(uint32_t index) const
+{
+ ASSERT(index < m_list->mNumberBuffers);
+ if (index < m_list->mNumberBuffers)
+ return &m_list->mBuffers[index];
+ return nullptr;
+}
+
+}
Added: trunk/Source/WebCore/platform/audio/WebAudioBufferList.h (0 => 211959)
--- trunk/Source/WebCore/platform/audio/WebAudioBufferList.h (rev 0)
+++ trunk/Source/WebCore/platform/audio/WebAudioBufferList.h 2017-02-09 16:33:06 UTC (rev 211959)
@@ -0,0 +1,68 @@
+/*
+ * 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 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. OR 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
+
+#include "PlatformAudioData.h"
+#include <wtf/IteratorRange.h>
+#include <wtf/RetainPtr.h>
+#include <wtf/Vector.h>
+
+struct AudioBuffer;
+struct AudioBufferList;
+typedef struct OpaqueCMBlockBuffer *CMBlockBufferRef;
+typedef struct opaqueCMSampleBuffer *CMSampleBufferRef;
+
+namespace WebCore {
+
+class CAAudioStreamDescription;
+
+class WebAudioBufferList : public PlatformAudioData {
+public:
+ WebAudioBufferList(const CAAudioStreamDescription&);
+ WebAudioBufferList(const CAAudioStreamDescription&, uint32_t sampleCount);
+ WebAudioBufferList(const CAAudioStreamDescription&, CMSampleBufferRef);
+
+ AudioBufferList* list() const { return m_list.get(); }
+ operator AudioBufferList&() const { return *m_list; }
+
+ uint32_t bufferCount() const;
+ AudioBuffer* buffer(uint32_t index) const;
+ WTF::IteratorRange<AudioBuffer*> buffers() const;
+
+private:
+ Kind kind() const { return Kind::WebAudioBufferList; }
+
+ size_t m_listBufferSize { 0 };
+ std::unique_ptr<AudioBufferList> m_list;
+ RetainPtr<CMBlockBufferRef> m_blockBuffer;
+ Vector<uint8_t> m_flatBuffer;
+};
+
+}
+
+SPECIALIZE_TYPE_TRAITS_BEGIN(WebCore::WebAudioBufferList)
+static bool isType(const WebCore::PlatformAudioData& data) { return data.kind() == WebCore::PlatformAudioData::Kind::WebAudioBufferList; }
+SPECIALIZE_TYPE_TRAITS_END()
Modified: trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.cpp (211958 => 211959)
--- trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.cpp 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.cpp 2017-02-09 16:33:06 UTC (rev 211959)
@@ -47,24 +47,9 @@
m_internalFormat = std::make_unique<CAAudioStreamDescription>(format);
m_sampleCapacity = maximumSampleCount;
- m_sampleCount = 0;
m_maxBufferSizePerChannel = maximumSampleCount * format.bytesPerFrame() / format.numberOfChannelStreams();
ASSERT(format.sampleRate() >= 0);
-
- size_t bufferSize = format.numberOfChannelStreams() * m_maxBufferSizePerChannel;
- ASSERT(bufferSize <= SIZE_MAX);
- if (bufferSize > SIZE_MAX)
- return;
-
- m_bufferListBaseSize = audioBufferListSizeForStream(format);
- ASSERT(m_bufferListBaseSize <= SIZE_MAX);
- if (m_bufferListBaseSize > SIZE_MAX)
- return;
-
- size_t allocSize = m_bufferListBaseSize + bufferSize;
- m_bufferList = std::unique_ptr<AudioBufferList>(static_cast<AudioBufferList*>(::operator new (allocSize)));
-
reset();
}
@@ -138,29 +123,29 @@
m_sampleCount = frameCount;
- AudioBufferList& sourceBuffer = source.bufferList();
- for (uint32_t i = 0; i < m_bufferList->mNumberBuffers; i++) {
+ WebAudioBufferList& sourceBuffer = source.bufferList();
+ for (uint32_t i = 0; i < m_bufferList->bufferCount(); i++) {
switch (m_internalFormat->format()) {
case AudioStreamDescription::Int16: {
- int16_t* destination = static_cast<int16_t*>(m_bufferList->mBuffers[i].mData);
- int16_t* source = static_cast<int16_t*>(sourceBuffer.mBuffers[i].mData);
+ int16_t* destination = static_cast<int16_t*>(m_bufferList->buffer(i)->mData);
+ int16_t* source = static_cast<int16_t*>(sourceBuffer.buffer(i)->mData);
for (size_t i = 0; i < frameCount; i++)
destination[i] += source[i];
break;
}
case AudioStreamDescription::Int32: {
- int32_t* destination = static_cast<int32_t*>(m_bufferList->mBuffers[i].mData);
- vDSP_vaddi(destination, 1, reinterpret_cast<int32_t*>(sourceBuffer.mBuffers[i].mData), 1, destination, 1, frameCount);
+ int32_t* destination = static_cast<int32_t*>(m_bufferList->buffer(i)->mData);
+ vDSP_vaddi(destination, 1, reinterpret_cast<int32_t*>(sourceBuffer.buffer(i)->mData), 1, destination, 1, frameCount);
break;
}
case AudioStreamDescription::Float32: {
- float* destination = static_cast<float*>(m_bufferList->mBuffers[i].mData);
- vDSP_vadd(destination, 1, reinterpret_cast<float*>(sourceBuffer.mBuffers[i].mData), 1, destination, 1, frameCount);
+ float* destination = static_cast<float*>(m_bufferList->buffer(i)->mData);
+ vDSP_vadd(destination, 1, reinterpret_cast<float*>(sourceBuffer.buffer(i)->mData), 1, destination, 1, frameCount);
break;
}
case AudioStreamDescription::Float64: {
- double* destination = static_cast<double*>(m_bufferList->mBuffers[i].mData);
- vDSP_vaddD(destination, 1, reinterpret_cast<double*>(sourceBuffer.mBuffers[i].mData), 1, destination, 1, frameCount);
+ double* destination = static_cast<double*>(m_bufferList->buffer(i)->mData);
+ vDSP_vaddD(destination, 1, reinterpret_cast<double*>(sourceBuffer.buffer(i)->mData), 1, destination, 1, frameCount);
break;
}
case AudioStreamDescription::None:
@@ -187,9 +172,9 @@
m_sampleCount = frameCount;
- for (uint32_t i = 0; i < m_bufferList->mNumberBuffers; i++) {
- uint8_t* sourceData = static_cast<uint8_t*>(source.bufferList().mBuffers[i].mData);
- uint8_t* destination = static_cast<uint8_t*>(m_bufferList->mBuffers[i].mData);
+ for (uint32_t i = 0; i < m_bufferList->bufferCount(); i++) {
+ uint8_t* sourceData = static_cast<uint8_t*>(source.bufferList().buffer(i)->mData);
+ uint8_t* destination = static_cast<uint8_t*>(m_bufferList->buffer(i)->mData);
memcpy(destination, sourceData, frameCount * m_internalFormat->bytesPerPacket());
}
@@ -200,11 +185,11 @@
{
if (frameCount > m_sampleCount)
return kAudio_ParamError;
- if (buffer.mNumberBuffers > m_bufferList->mNumberBuffers)
+ if (buffer.mNumberBuffers > m_bufferList->bufferCount())
return kAudio_ParamError;
for (uint32_t i = 0; i < buffer.mNumberBuffers; i++) {
- uint8_t* sourceData = static_cast<uint8_t*>(m_bufferList->mBuffers[i].mData);
+ uint8_t* sourceData = static_cast<uint8_t*>(m_bufferList->buffer(i)->mData);
uint8_t* destination = static_cast<uint8_t*>(buffer.mBuffers[i].mData);
memcpy(destination, sourceData, frameCount * m_internalFormat->bytesPerPacket());
}
@@ -218,15 +203,7 @@
m_timestamp = 0;
m_hostTime = -1;
- uint8_t* data = "" + m_bufferListBaseSize;
- m_bufferList->mNumberBuffers = m_internalFormat->numberOfChannelStreams();
- for (uint32_t i = 0; i < m_bufferList->mNumberBuffers; ++i) {
- auto& buffer = m_bufferList->mBuffers[i];
- buffer.mData = data;
- buffer.mDataByteSize = m_maxBufferSizePerChannel;
- buffer.mNumberChannels = m_internalFormat->numberOfInterleavedChannels();
- data = "" + m_maxBufferSizePerChannel;
- }
+ m_bufferList = std::make_unique<WebAudioBufferList>(*m_internalFormat, m_maxBufferSizePerChannel);
}
void AudioSampleBufferList::zero()
@@ -277,22 +254,20 @@
m_converterInputBytesPerPacket = inputFormat.mBytesPerPacket;
m_converterInputBuffer = &source;
- auto* outputData = m_bufferList.get();
-
#if !LOG_DISABLED
AudioStreamBasicDescription outputFormat;
propertyDataSize = sizeof(outputFormat);
AudioConverterGetProperty(converter, kAudioConverterCurrentOutputStreamDescription, &propertyDataSize, &outputFormat);
- ASSERT(outputFormat.mChannelsPerFrame == outputData->mNumberBuffers);
- for (uint32_t i = 0; i < outputData->mNumberBuffers; ++i) {
- ASSERT(outputData->mBuffers[i].mData);
- ASSERT(outputData->mBuffers[i].mDataByteSize);
+ ASSERT(outputFormat.mChannelsPerFrame == m_bufferList->bufferCount());
+ for (uint32_t i = 0; i < m_bufferList->bufferCount(); ++i) {
+ ASSERT(m_bufferList->buffer(i)->mData);
+ ASSERT(m_bufferList->buffer(i)->mDataByteSize);
}
#endif
UInt32 samplesConverted = static_cast<UInt32>(m_sampleCapacity);
- OSStatus err = AudioConverterFillComplexBuffer(converter, audioConverterCallback, this, &samplesConverted, outputData, nullptr);
+ OSStatus err = AudioConverterFillComplexBuffer(converter, audioConverterCallback, this, &samplesConverted, m_bufferList->list(), nullptr);
if (err) {
LOG_ERROR("AudioSampleBufferList::copyFrom(%p) AudioConverterFillComplexBuffer returned error %d (%.4s)", this, (int)err, (char*)&err);
m_sampleCount = std::min(m_sampleCapacity, static_cast<size_t>(samplesConverted));
@@ -312,7 +287,7 @@
OSStatus AudioSampleBufferList::copyFrom(CARingBuffer& ringBuffer, size_t sampleCount, uint64_t startFrame, CARingBuffer::FetchMode mode)
{
reset();
- if (ringBuffer.fetch(&bufferList(), sampleCount, startFrame, mode) != CARingBuffer::Ok)
+ if (ringBuffer.fetch(bufferList().list(), sampleCount, startFrame, mode) != CARingBuffer::Ok)
return kAudio_ParamError;
m_sampleCount = sampleCount;
@@ -319,22 +294,6 @@
return 0;
}
-void AudioSampleBufferList::configureBufferListForStream(AudioBufferList& bufferList, const CAAudioStreamDescription& format, uint8_t* bufferData, size_t sampleCount)
-{
- size_t bufferCount = format.numberOfChannelStreams();
- size_t channelCount = format.numberOfInterleavedChannels();
- size_t bytesPerChannel = sampleCount * format.bytesPerFrame();
-
- bufferList.mNumberBuffers = bufferCount;
- for (unsigned i = 0; i < bufferCount; ++i) {
- bufferList.mBuffers[i].mNumberChannels = channelCount;
- bufferList.mBuffers[i].mDataByteSize = bytesPerChannel;
- bufferList.mBuffers[i].mData = bufferData;
- if (bufferData)
- bufferData = bufferData + bytesPerChannel;
- }
-}
-
} // namespace WebCore
#endif // ENABLE(MEDIA_STREAM)
Modified: trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.h (211958 => 211959)
--- trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.h 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.h 2017-02-09 16:33:06 UTC (rev 211959)
@@ -28,6 +28,7 @@
#if ENABLE(MEDIA_STREAM)
#include "CARingBuffer.h"
+#include "WebAudioBufferList.h"
#include <CoreAudio/CoreAudioTypes.h>
#include <wtf/Lock.h>
#include <wtf/RefCounted.h>
@@ -44,7 +45,6 @@
~AudioSampleBufferList();
- static void configureBufferListForStream(AudioBufferList&, const CAAudioStreamDescription&, uint8_t*, size_t);
static inline size_t audioBufferListSizeForStream(const CAAudioStreamDescription&);
static void applyGain(AudioBufferList&, float, AudioStreamDescription::PCMFormat);
@@ -60,7 +60,7 @@
OSStatus copyTo(AudioBufferList&, size_t count = SIZE_MAX);
const AudioStreamBasicDescription& streamDescription() const { return m_internalFormat->streamDescription(); }
- AudioBufferList& bufferList() const { return *m_bufferList.get(); }
+ WebAudioBufferList& bufferList() const { return *m_bufferList; }
uint32_t sampleCapacity() const { return m_sampleCapacity; }
uint32_t sampleCount() const { return m_sampleCount; }
@@ -93,7 +93,7 @@
size_t m_sampleCapacity { 0 };
size_t m_maxBufferSizePerChannel { 0 };
size_t m_bufferListBaseSize { 0 };
- std::unique_ptr<AudioBufferList> m_bufferList;
+ std::unique_ptr<WebAudioBufferList> m_bufferList;
};
inline size_t AudioSampleBufferList::audioBufferListSizeForStream(const CAAudioStreamDescription& description)
Modified: trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp (211958 => 211959)
--- trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp 2017-02-09 16:33:06 UTC (rev 211959)
@@ -152,7 +152,7 @@
if (err)
return;
- sampleBufferList = &m_scratchBuffer->bufferList();
+ sampleBufferList = m_scratchBuffer->bufferList().list();
} else
sampleBufferList = &bufferList;
@@ -192,34 +192,16 @@
ASSERT_UNUSED(sampleDescription, *m_inputDescription == sampleDescription);
ASSERT(m_ringBuffer);
-
- size_t bufferSize = AudioSampleBufferList::audioBufferListSizeForStream(*m_inputDescription.get());
- uint8_t bufferData[bufferSize];
- AudioBufferList* bufferList = reinterpret_cast<AudioBufferList*>(bufferData);
- bufferList->mNumberBuffers = m_inputDescription->numberOfInterleavedChannels();
-
- CMBlockBufferRef buffer = nullptr;
- OSStatus err = CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampleBuffer, nullptr, bufferList, bufferSize, kCFAllocatorSystemDefault, kCFAllocatorSystemDefault, kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, &buffer);
- if (err) {
- LOG_ERROR("AudioSampleDataSource::pushSamples(%p) - CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer returned error %d (%.4s)", this, (int)err, (char*)&err);
- return;
- }
-
- pushSamplesInternal(*bufferList, toMediaTime(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)), CMSampleBufferGetNumSamples(sampleBuffer));
+
+ WebAudioBufferList list(*m_inputDescription, sampleBuffer);
+ pushSamplesInternal(list, toMediaTime(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)), CMSampleBufferGetNumSamples(sampleBuffer));
}
-void AudioSampleDataSource::pushSamples(const AudioStreamBasicDescription& sampleDescription, const MediaTime& sampleTime, void* audioData, size_t sampleCount)
+void AudioSampleDataSource::pushSamples(const MediaTime& sampleTime, PlatformAudioData& audioData, size_t sampleCount)
{
std::unique_lock<Lock> lock(m_lock, std::try_to_lock);
- ASSERT(*m_inputDescription == sampleDescription);
-
- CAAudioStreamDescription description(sampleDescription);
- size_t bufferSize = AudioSampleBufferList::audioBufferListSizeForStream(description);
- uint8_t bufferData[bufferSize];
- AudioBufferList* bufferList = reinterpret_cast<AudioBufferList*>(bufferData);
-
- AudioSampleBufferList::configureBufferListForStream(*bufferList, description, reinterpret_cast<uint8_t*>(audioData), sampleCount);
- pushSamplesInternal(*bufferList, sampleTime, sampleCount);
+ ASSERT(is<WebAudioBufferList>(audioData));
+ pushSamplesInternal(downcast<WebAudioBufferList>(audioData), sampleTime, sampleCount);
}
bool AudioSampleDataSource::pullSamplesInternal(AudioBufferList& buffer, size_t& sampleCount, uint64_t timeStamp, double /*hostTime*/, PullMode mode)
Modified: trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.h (211958 => 211959)
--- trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.h 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.h 2017-02-09 16:33:06 UTC (rev 211959)
@@ -52,7 +52,7 @@
OSStatus setInputFormat(const CAAudioStreamDescription&);
OSStatus setOutputFormat(const CAAudioStreamDescription&);
- void pushSamples(const AudioStreamBasicDescription&, const MediaTime&, void*, size_t);
+ void pushSamples(const MediaTime&, PlatformAudioData&, size_t);
void pushSamples(const AudioStreamBasicDescription&, CMSampleBufferRef);
enum PullMode { Copy, Mix };
Modified: trunk/Source/WebCore/platform/mediastream/RealtimeMediaSource.cpp (211958 => 211959)
--- trunk/Source/WebCore/platform/mediastream/RealtimeMediaSource.cpp 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/mediastream/RealtimeMediaSource.cpp 2017-02-09 16:33:06 UTC (rev 211959)
@@ -119,7 +119,7 @@
observer->videoSampleAvailable(mediaSample);
}
-void RealtimeMediaSource::audioSamplesAvailable(const MediaTime& time, void* audioData, const AudioStreamDescription& description, size_t numberOfFrames)
+void RealtimeMediaSource::audioSamplesAvailable(const MediaTime& time, PlatformAudioData& audioData, const AudioStreamDescription& description, size_t numberOfFrames)
{
for (const auto& observer : m_observers)
observer->audioSamplesAvailable(time, audioData, description, numberOfFrames);
Modified: trunk/Source/WebCore/platform/mediastream/RealtimeMediaSource.h (211958 => 211959)
--- trunk/Source/WebCore/platform/mediastream/RealtimeMediaSource.h 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/mediastream/RealtimeMediaSource.h 2017-02-09 16:33:06 UTC (rev 211959)
@@ -57,6 +57,7 @@
class FloatRect;
class GraphicsContext;
class MediaStreamPrivate;
+class PlatformAudioData;
class RealtimeMediaSourceSettings;
class RealtimeMediaSource : public RefCounted<RealtimeMediaSource> {
@@ -77,7 +78,7 @@
virtual void videoSampleAvailable(MediaSample&) { }
// May be called on a background thread.
- virtual void audioSamplesAvailable(const MediaTime&, void* /*audioData*/, const AudioStreamDescription&, size_t /*numberOfFrames*/) { }
+ virtual void audioSamplesAvailable(const MediaTime&, PlatformAudioData&, const AudioStreamDescription&, size_t /*numberOfFrames*/) { }
};
virtual ~RealtimeMediaSource() { }
@@ -109,7 +110,7 @@
virtual void settingsDidChange();
void videoSampleAvailable(MediaSample&);
- void audioSamplesAvailable(const MediaTime&, void*, const AudioStreamDescription&, size_t);
+ void audioSamplesAvailable(const MediaTime&, PlatformAudioData&, const AudioStreamDescription&, size_t);
bool stopped() const { return m_stopped; }
Modified: trunk/Source/WebCore/platform/mediastream/mac/AVAudioCaptureSource.h (211958 => 211959)
--- trunk/Source/WebCore/platform/mediastream/mac/AVAudioCaptureSource.h 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/mediastream/mac/AVAudioCaptureSource.h 2017-02-09 16:33:06 UTC (rev 211959)
@@ -39,6 +39,7 @@
namespace WebCore {
+class WebAudioBufferList;
class WebAudioSourceProviderAVFObjC;
class AVAudioCaptureSource : public AVMediaCaptureSource, public AudioCaptureSourceProviderObjC {
@@ -66,8 +67,7 @@
AudioSourceProvider* audioSourceProvider() override;
RetainPtr<AVCaptureConnection> m_audioConnection;
- size_t m_listBufferSize { 0 };
- std::unique_ptr<AudioBufferList> m_list;
+ std::unique_ptr<WebAudioBufferList> m_list;
RefPtr<WebAudioSourceProviderAVFObjC> m_audioSourceProvider;
std::unique_ptr<CAAudioStreamDescription> m_inputDescription;
Modified: trunk/Source/WebCore/platform/mediastream/mac/AVAudioCaptureSource.mm (211958 => 211959)
--- trunk/Source/WebCore/platform/mediastream/mac/AVAudioCaptureSource.mm 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/mediastream/mac/AVAudioCaptureSource.mm 2017-02-09 16:33:06 UTC (rev 211959)
@@ -193,10 +193,6 @@
const AudioStreamBasicDescription* streamDescription = CMAudioFormatDescriptionGetStreamBasicDescription(formatDescription);
if (!m_inputDescription || *m_inputDescription != *streamDescription) {
m_inputDescription = std::make_unique<CAAudioStreamDescription>(*streamDescription);
- m_listBufferSize = AudioSampleBufferList::audioBufferListSizeForStream(*m_inputDescription.get());
- m_list = std::unique_ptr<AudioBufferList>(static_cast<AudioBufferList*>(::operator new (m_listBufferSize)));
- memset(m_list.get(), 0, m_listBufferSize);
- m_list->mNumberBuffers = m_inputDescription->numberOfChannelStreams();
if (!m_observers.isEmpty()) {
for (auto& observer : m_observers)
@@ -204,13 +200,8 @@
}
}
- CMItemCount frameCount = CMSampleBufferGetNumSamples(sampleBuffer);
- CMBlockBufferRef buffer = nil;
- OSStatus err = CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampleBuffer, nullptr, m_list.get(), m_listBufferSize, kCFAllocatorSystemDefault, kCFAllocatorSystemDefault, kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, &buffer);
- if (!err)
- audioSamplesAvailable(toMediaTime(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)), m_list->mBuffers[0].mData, CAAudioStreamDescription(*streamDescription), frameCount);
- else
- LOG_ERROR("AVAudioCaptureSource::captureOutputDidOutputSampleBufferFromConnection(%p) - CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer returned error %d (%.4s)", this, (int)err, (char*)&err);
+ m_list = std::make_unique<WebAudioBufferList>(*m_inputDescription, sampleBuffer);
+ audioSamplesAvailable(toMediaTime(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)), *m_list, CAAudioStreamDescription(*streamDescription), CMSampleBufferGetNumSamples(sampleBuffer));
if (m_observers.isEmpty())
return;
Modified: trunk/Source/WebCore/platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.cpp (211958 => 211959)
--- trunk/Source/WebCore/platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.cpp 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.cpp 2017-02-09 16:33:06 UTC (rev 211959)
@@ -178,7 +178,7 @@
return err;
}
-void AudioTrackPrivateMediaStreamCocoa::audioSamplesAvailable(const MediaTime& sampleTime, void* audioData, const AudioStreamDescription& description, size_t sampleCount)
+void AudioTrackPrivateMediaStreamCocoa::audioSamplesAvailable(const MediaTime& sampleTime, PlatformAudioData& audioData, const AudioStreamDescription& description, size_t sampleCount)
{
ASSERT(description.platformDescription().type == PlatformDescription::CAAudioStreamBasicType);
@@ -215,7 +215,7 @@
m_dataSource->setVolume(m_volume);
}
- m_dataSource->pushSamples(m_inputDescription->streamDescription(), sampleTime, audioData, sampleCount);
+ m_dataSource->pushSamples(sampleTime, audioData, sampleCount);
if (m_autoPlay)
playInternal();
Modified: trunk/Source/WebCore/platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.h (211958 => 211959)
--- trunk/Source/WebCore/platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.h 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/mediastream/mac/AudioTrackPrivateMediaStreamCocoa.h 2017-02-09 16:33:06 UTC (rev 211959)
@@ -66,7 +66,7 @@
void sourceMutedChanged() final { }
void sourceSettingsChanged() final { }
bool preventSourceFromStopping() final { return false; }
- void audioSamplesAvailable(const MediaTime&, void*, const AudioStreamDescription&, size_t) final;
+ void audioSamplesAvailable(const MediaTime&, PlatformAudioData&, const AudioStreamDescription&, size_t) final;
static OSStatus inputProc(void*, AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32 inBusNumber, UInt32 numberOfFrames, AudioBufferList*);
OSStatus render(UInt32 sampleCount, AudioBufferList&, UInt32 inBusNumber, const AudioTimeStamp&, AudioUnitRenderActionFlags&);
Modified: trunk/Source/WebCore/platform/mediastream/mac/MockRealtimeAudioSourceMac.h (211958 => 211959)
--- trunk/Source/WebCore/platform/mediastream/mac/MockRealtimeAudioSourceMac.h 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/mediastream/mac/MockRealtimeAudioSourceMac.h 2017-02-09 16:33:06 UTC (rev 211959)
@@ -38,12 +38,12 @@
#include <CoreAudio/CoreAudioTypes.h>
OBJC_CLASS AVAudioPCMBuffer;
-typedef struct AudioBufferList AudioBufferList;
typedef struct OpaqueCMClock* CMClockRef;
typedef const struct opaqueCMFormatDescription* CMFormatDescriptionRef;
namespace WebCore {
+class WebAudioBufferList;
class WebAudioSourceProviderAVFObjC;
class MockRealtimeAudioSourceMac final : public MockRealtimeAudioSource, public AudioCaptureSourceProviderObjC {
@@ -68,7 +68,7 @@
AudioSourceProvider* audioSourceProvider() final;
size_t m_audioBufferListBufferSize { 0 };
- std::unique_ptr<AudioBufferList> m_audioBufferList;
+ std::unique_ptr<WebAudioBufferList> m_audioBufferList;
uint32_t m_maximiumFrameCount;
uint32_t m_sampleRate { 44100 };
Modified: trunk/Source/WebCore/platform/mediastream/mac/MockRealtimeAudioSourceMac.mm (211958 => 211959)
--- trunk/Source/WebCore/platform/mediastream/mac/MockRealtimeAudioSourceMac.mm 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/mediastream/mac/MockRealtimeAudioSourceMac.mm 2017-02-09 16:33:06 UTC (rev 211959)
@@ -38,6 +38,7 @@
#import "MediaSampleAVFObjC.h"
#import "NotImplemented.h"
#import "RealtimeMediaSourceSettings.h"
+#import "WebAudioBufferList.h"
#import "WebAudioSourceProviderAVFObjC.h"
#import <AVFoundation/AVAudioBuffer.h>
#import <AudioToolbox/AudioConverter.h>
@@ -102,7 +103,7 @@
CMTime startTime = CMTimeMake(m_bytesEmitted, m_sampleRate);
m_bytesEmitted += frameCount;
- audioSamplesAvailable(toMediaTime(startTime), m_audioBufferList->mBuffers[0].mData, CAAudioStreamDescription(m_streamFormat), frameCount);
+ audioSamplesAvailable(toMediaTime(startTime), *m_audioBufferList, CAAudioStreamDescription(m_streamFormat), frameCount);
CMSampleBufferRef sampleBuffer;
OSStatus result = CMAudioSampleBufferCreateWithPacketDescriptions(nullptr, nullptr, true, nullptr, nullptr, m_formatDescription.get(), frameCount, startTime, nullptr, &sampleBuffer);
@@ -113,7 +114,7 @@
return;
auto buffer = adoptCF(sampleBuffer);
- result = CMSampleBufferSetDataBufferFromAudioBufferList(sampleBuffer, kCFAllocatorDefault, kCFAllocatorDefault, 0, m_audioBufferList.get());
+ result = CMSampleBufferSetDataBufferFromAudioBufferList(sampleBuffer, kCFAllocatorDefault, kCFAllocatorDefault, 0, m_audioBufferList->list());
ASSERT(!result);
result = CMSampleBufferSetDataReady(sampleBuffer);
@@ -130,34 +131,14 @@
const int bytesPerFloat = sizeof(Float32);
const int bitsPerByte = 8;
- int channelCount = 1;
- m_streamFormat = { };
- m_streamFormat.mSampleRate = m_sampleRate;
- m_streamFormat.mFormatID = kAudioFormatLinearPCM;
- m_streamFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
- m_streamFormat.mBytesPerPacket = bytesPerFloat * channelCount;
- m_streamFormat.mFramesPerPacket = 1;
- m_streamFormat.mBytesPerFrame = bytesPerFloat * channelCount;
- m_streamFormat.mChannelsPerFrame = channelCount;
- m_streamFormat.mBitsPerChannel = bitsPerByte * bytesPerFloat;
+ const int channelCount = 2;
+ const bool isFloat = true;
+ const bool isBigEndian = false;
+ const bool isNonInterleaved = true;
+ FillOutASBDForLPCM(m_streamFormat, m_sampleRate, channelCount, bitsPerByte * bytesPerFloat, bitsPerByte * bytesPerFloat, isFloat, isBigEndian, isNonInterleaved);
- // AudioBufferList is a variable-length struct, so create on the heap with a generic new() operator
- // with a custom size, and initialize the struct manually.
- uint32_t bufferDataSize = m_streamFormat.mBytesPerFrame * m_maximiumFrameCount;
- uint32_t baseSize = AudioSampleBufferList::audioBufferListSizeForStream(m_streamFormat);
+ m_audioBufferList = std::make_unique<WebAudioBufferList>(m_streamFormat, m_streamFormat.mBytesPerFrame * m_maximiumFrameCount);
- uint64_t bufferListSize = baseSize + bufferDataSize;
- ASSERT(bufferListSize <= SIZE_MAX);
- if (bufferListSize > SIZE_MAX)
- return;
-
- m_audioBufferListBufferSize = static_cast<size_t>(bufferListSize);
- m_audioBufferList = std::unique_ptr<AudioBufferList>(static_cast<AudioBufferList*>(::operator new (m_audioBufferListBufferSize)));
- memset(m_audioBufferList.get(), 0, m_audioBufferListBufferSize);
-
- uint8_t* bufferData = reinterpret_cast<uint8_t*>(m_audioBufferList.get()) + baseSize;
- AudioSampleBufferList::configureBufferListForStream(*m_audioBufferList.get(), m_streamFormat, bufferData, bufferDataSize);
-
CMFormatDescriptionRef formatDescription;
CMAudioFormatDescriptionCreate(NULL, &m_streamFormat, 0, NULL, 0, NULL, NULL, &formatDescription);
m_formatDescription = adoptCF(formatDescription);
@@ -178,40 +159,42 @@
uint32_t totalFrameCount = alignTo16Bytes(delta * m_sampleRate);
uint32_t frameCount = std::min(totalFrameCount, m_maximiumFrameCount);
double elapsed = elapsedTime();
+
while (frameCount) {
- float *buffer = static_cast<float *>(m_audioBufferList->mBuffers[0].mData);
- for (uint32_t frame = 0; frame < frameCount; ++frame) {
- int phase = fmod(elapsed, 2) * 15;
- double increment = 0;
- bool silent = true;
+ for (auto& audioBuffer : m_audioBufferList->buffers()) {
+ audioBuffer.mDataByteSize = frameCount * m_streamFormat.mBytesPerFrame;
+ float *buffer = static_cast<float *>(audioBuffer.mData);
+ for (uint32_t frame = 0; frame < frameCount; ++frame) {
+ int phase = fmod(elapsed, 2) * 15;
+ double increment = 0;
+ bool silent = true;
- switch (phase) {
- case 0:
- case 14: {
- int index = fmod(elapsed, 1) * 2;
- increment = tau * frequencies[index] / m_sampleRate;
- silent = false;
- break;
- }
- default:
- break;
- }
+ switch (phase) {
+ case 0:
+ case 14: {
+ int index = fmod(elapsed, 1) * 2;
+ increment = tau * frequencies[index] / m_sampleRate;
+ silent = false;
+ break;
+ }
+ default:
+ break;
+ }
- if (silent) {
- buffer[frame] = 0;
- continue;
- }
+ if (silent) {
+ buffer[frame] = 0;
+ continue;
+ }
- float tone = sin(theta) * 0.25;
- buffer[frame] = tone;
+ float tone = sin(theta) * 0.25;
+ buffer[frame] = tone;
theta += increment;
- if (theta > tau)
- theta -= tau;
- elapsed += 1 / m_sampleRate;
+ if (theta > tau)
+ theta -= tau;
+ elapsed += 1 / m_sampleRate;
+ }
}
-
- m_audioBufferList->mBuffers[0].mDataByteSize = frameCount * sizeof(float);
emitSampleBuffers(frameCount);
totalFrameCount -= frameCount;
frameCount = std::min(totalFrameCount, m_maximiumFrameCount);
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp (211958 => 211959)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-02-09 16:33:06 UTC (rev 211959)
@@ -35,7 +35,7 @@
namespace WebCore {
-void RealtimeOutgoingAudioSource::audioSamplesAvailable(const MediaTime&, void*, const AudioStreamDescription&, size_t)
+void RealtimeOutgoingAudioSource::audioSamplesAvailable(const MediaTime&, PlatformAudioData&, const AudioStreamDescription&, size_t)
{
notImplemented();
}
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h (211958 => 211959)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h 2017-02-09 16:32:23 UTC (rev 211958)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h 2017-02-09 16:33:06 UTC (rev 211959)
@@ -66,7 +66,7 @@
void sourceMutedChanged() final { }
void sourceSettingsChanged() final { }
bool preventSourceFromStopping() final { return false; }
- void audioSamplesAvailable(const MediaTime&, void*, const AudioStreamDescription&, size_t) final;
+ void audioSamplesAvailable(const MediaTime&, PlatformAudioData&, const AudioStreamDescription&, size_t) final;
void convertAndSendMonoSamples();
void convertAndSendStereoSamples();
