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<title>[211610] trunk/Source/WebCore</title>
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<div id="msg">
<dl class="meta">
<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/211610">211610</a></dd>
<dt>Author</dt> <dd>ryanhaddad@apple.com</dd>
<dt>Date</dt> <dd>2017-02-02 16:35:30 -0800 (Thu, 02 Feb 2017)</dd>
</dl>
<h3>Log Message</h3>
<pre>Unreviewed, rolling out <a href="http://trac.webkit.org/projects/webkit/changeset/211596">r211596</a> and <a href="http://trac.webkit.org/projects/webkit/changeset/211605">r211605</a>.
https://bugs.webkit.org/show_bug.cgi?id=167767
This change broke certain build configurations (Requested by
ryanhaddad on #webkit).
Reverted changesets:
"[Mac] Add classes to manage audio samples"
https://bugs.webkit.org/show_bug.cgi?id=167739
http://trac.webkit.org/changeset/211596
"Unreviewed speculative build fix."
http://trac.webkit.org/changeset/211605
Patch by Commit Queue <commit-queue@webkit.org> on 2017-02-02</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkSourceWebCoreChangeLog">trunk/Source/WebCore/ChangeLog</a></li>
<li><a href="#trunkSourceWebCoreWebCorexcodeprojprojectpbxproj">trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj</a></li>
</ul>
<h3>Removed Paths</h3>
<ul>
<li><a href="#trunkSourceWebCoreplatformaudiomacAudioSampleBufferListcpp">trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.cpp</a></li>
<li><a href="#trunkSourceWebCoreplatformaudiomacAudioSampleBufferListh">trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.h</a></li>
<li><a href="#trunkSourceWebCoreplatformaudiomacAudioSampleDataSourcecpp">trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp</a></li>
<li><a href="#trunkSourceWebCoreplatformaudiomacAudioSampleDataSourceh">trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.h</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkSourceWebCoreChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/ChangeLog (211609 => 211610)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/ChangeLog 2017-02-03 00:26:00 UTC (rev 211609)
+++ trunk/Source/WebCore/ChangeLog 2017-02-03 00:35:30 UTC (rev 211610)
</span><span class="lines">@@ -1,3 +1,20 @@
</span><ins>+2017-02-02 Commit Queue <commit-queue@webkit.org>
+
+ Unreviewed, rolling out r211596 and r211605.
+ https://bugs.webkit.org/show_bug.cgi?id=167767
+
+ This change broke certain build configurations (Requested by
+ ryanhaddad on #webkit).
+
+ Reverted changesets:
+
+ "[Mac] Add classes to manage audio samples"
+ https://bugs.webkit.org/show_bug.cgi?id=167739
+ http://trac.webkit.org/changeset/211596
+
+ "Unreviewed speculative build fix."
+ http://trac.webkit.org/changeset/211605
+
</ins><span class="cx"> 2017-02-02 Ryan Haddad <ryanhaddad@apple.com>
</span><span class="cx">
</span><span class="cx"> Unreviewed speculative build fix.
</span></span></pre></div>
<a id="trunkSourceWebCoreWebCorexcodeprojprojectpbxproj"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj (211609 => 211610)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2017-02-03 00:26:00 UTC (rev 211609)
+++ trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2017-02-03 00:35:30 UTC (rev 211610)
</span><span class="lines">@@ -142,10 +142,6 @@
</span><span class="cx"> 073794FE19F5864E00E5A045 /* RTCNotifiersMock.h in Headers */ = {isa = PBXBuildFile; fileRef = 073794F819F5864E00E5A045 /* RTCNotifiersMock.h */; };
</span><span class="cx"> 07394EC81BAB2CCD00BE99CD /* MediaDevicesRequest.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 07394EC71BAB2CCD00BE99CD /* MediaDevicesRequest.cpp */; };
</span><span class="cx"> 07394ECA1BAB2CD700BE99CD /* MediaDevicesRequest.h in Headers */ = {isa = PBXBuildFile; fileRef = 07394EC91BAB2CD700BE99CD /* MediaDevicesRequest.h */; settings = {ATTRIBUTES = (Private, ); }; };
</span><del>- 073B87661E4385AC0071C0EC /* AudioSampleBufferList.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 073B87621E43859D0071C0EC /* AudioSampleBufferList.cpp */; };
- 073B87671E4385AC0071C0EC /* AudioSampleBufferList.h in Headers */ = {isa = PBXBuildFile; fileRef = 073B87631E43859D0071C0EC /* AudioSampleBufferList.h */; };
- 073B87681E4385AC0071C0EC /* AudioSampleDataSource.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 073B87641E43859D0071C0EC /* AudioSampleDataSource.cpp */; };
- 073B87691E4385AC0071C0EC /* AudioSampleDataSource.h in Headers */ = {isa = PBXBuildFile; fileRef = 073B87651E43859D0071C0EC /* AudioSampleDataSource.h */; };
</del><span class="cx"> 073BE34017D17E01002BD431 /* JSNavigatorUserMedia.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 073BE33E17D17E01002BD431 /* JSNavigatorUserMedia.cpp */; };
</span><span class="cx"> 073BE34117D17E01002BD431 /* JSNavigatorUserMedia.h in Headers */ = {isa = PBXBuildFile; fileRef = 073BE33F17D17E01002BD431 /* JSNavigatorUserMedia.h */; settings = {ATTRIBUTES = (Private, ); }; };
</span><span class="cx"> 073BE34E17D180B2002BD431 /* RTCSessionDescriptionDescriptor.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 07221BAB17CF0AD400848E51 /* RTCSessionDescriptionDescriptor.cpp */; };
</span><span class="lines">@@ -7240,10 +7236,6 @@
</span><span class="cx"> 073B87561E40DCE50071C0EC /* AudioStreamDescription.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = AudioStreamDescription.h; sourceTree = "<group>"; };
</span><span class="cx"> 073B87571E40DCFD0071C0EC /* CAAudioStreamDescription.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = CAAudioStreamDescription.cpp; sourceTree = "<group>"; };
</span><span class="cx"> 073B87581E40DCFD0071C0EC /* CAAudioStreamDescription.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CAAudioStreamDescription.h; sourceTree = "<group>"; };
</span><del>- 073B87621E43859D0071C0EC /* AudioSampleBufferList.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = AudioSampleBufferList.cpp; sourceTree = "<group>"; };
- 073B87631E43859D0071C0EC /* AudioSampleBufferList.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = AudioSampleBufferList.h; sourceTree = "<group>"; };
- 073B87641E43859D0071C0EC /* AudioSampleDataSource.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = AudioSampleDataSource.cpp; sourceTree = "<group>"; };
- 073B87651E43859D0071C0EC /* AudioSampleDataSource.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = AudioSampleDataSource.h; sourceTree = "<group>"; };
</del><span class="cx"> 073BE33E17D17E01002BD431 /* JSNavigatorUserMedia.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSNavigatorUserMedia.cpp; sourceTree = "<group>"; };
</span><span class="cx"> 073BE33F17D17E01002BD431 /* JSNavigatorUserMedia.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = JSNavigatorUserMedia.h; sourceTree = "<group>"; };
</span><span class="cx"> 0744ECEB1E0C4AE5000D0944 /* MockRealtimeAudioSourceMac.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = MockRealtimeAudioSourceMac.h; sourceTree = "<group>"; };
</span><span class="lines">@@ -24868,6 +24860,7 @@
</span><span class="cx"> FD31604012B026A300C1A359 /* audio */ = {
</span><span class="cx"> isa = PBXGroup;
</span><span class="cx"> children = (
</span><ins>+ 073B87561E40DCE50071C0EC /* AudioStreamDescription.h */,
</ins><span class="cx"> CD669D651D232DF4004D1866 /* cocoa */,
</span><span class="cx"> CD0EEE0D14743E48003EAFA2 /* ios */,
</span><span class="cx"> FD3160B012B0270700C1A359 /* mac */,
</span><span class="lines">@@ -24894,7 +24887,6 @@
</span><span class="cx"> CDA79821170A22DC00D45C55 /* AudioSession.h */,
</span><span class="cx"> FD31605312B026F700C1A359 /* AudioSourceProvider.h */,
</span><span class="cx"> FD62F52D145898D80094B0ED /* AudioSourceProviderClient.h */,
</span><del>- 073B87561E40DCE50071C0EC /* AudioStreamDescription.h */,
</del><span class="cx"> FD31605412B026F700C1A359 /* AudioUtilities.cpp */,
</span><span class="cx"> FD31605512B026F700C1A359 /* AudioUtilities.h */,
</span><span class="cx"> FD31605612B026F700C1A359 /* Biquad.cpp */,
</span><span class="lines">@@ -24961,10 +24953,6 @@
</span><span class="cx"> FD3160B012B0270700C1A359 /* mac */ = {
</span><span class="cx"> isa = PBXGroup;
</span><span class="cx"> children = (
</span><del>- 073B87621E43859D0071C0EC /* AudioSampleBufferList.cpp */,
- 073B87631E43859D0071C0EC /* AudioSampleBufferList.h */,
- 073B87641E43859D0071C0EC /* AudioSampleDataSource.cpp */,
- 073B87651E43859D0071C0EC /* AudioSampleDataSource.h */,
</del><span class="cx"> FD3160B512B0272A00C1A359 /* AudioBusMac.mm */,
</span><span class="cx"> FD3160B612B0272A00C1A359 /* AudioDestinationMac.cpp */,
</span><span class="cx"> FD3160B712B0272A00C1A359 /* AudioDestinationMac.h */,
</span><span class="lines">@@ -25529,7 +25517,6 @@
</span><span class="cx"> 2D8FEBDD143E3EF70072502B /* CSSCrossfadeValue.h in Headers */,
</span><span class="cx"> AA21ECCD0ABF0FC6002B834C /* CSSCursorImageValue.h in Headers */,
</span><span class="cx"> 9444CBE41D8861990073A074 /* CSSCustomIdentValue.h in Headers */,
</span><del>- 073B87691E4385AC0071C0EC /* AudioSampleDataSource.h in Headers */,
</del><span class="cx"> BC779E141BB215BB00CAA8BF /* CSSCustomPropertyValue.h in Headers */,
</span><span class="cx"> 4A9CC81816BB9AC600EC645A /* CSSDefaultStyleSheets.h in Headers */,
</span><span class="cx"> 94476BDB1DFCAC0300690E23 /* CSSDeferredParser.h in Headers */,
</span><span class="lines">@@ -26711,7 +26698,6 @@
</span><span class="cx"> 6C4C96DF1AD4483500363F64 /* JSReadableByteStreamController.h in Headers */,
</span><span class="cx"> 7C4C96DD1AD4483500365A50 /* JSReadableStream.h in Headers */,
</span><span class="cx"> 6C4C96DF1AD4483500365A50 /* JSReadableStreamDefaultController.h in Headers */,
</span><del>- 073B87671E4385AC0071C0EC /* AudioSampleBufferList.h in Headers */,
</del><span class="cx"> 7C4C96DF1AD4483500365A50 /* JSReadableStreamDefaultReader.h in Headers */,
</span><span class="cx"> 4129DF861BB5B80C00322A16 /* JSReadableStreamPrivateConstructors.h in Headers */,
</span><span class="cx"> 7E4C96DD1AD4483500365A51 /* JSReadableStreamSource.h in Headers */,
</span><span class="lines">@@ -28398,6 +28384,7 @@
</span><span class="cx"> BE913D80181EF92400DCB09E /* TrackPrivateBase.h in Headers */,
</span><span class="cx"> FFAC30FE184FB145008C4F1E /* TrailingObjects.h in Headers */,
</span><span class="cx"> 516071321BD8308B00DBC4F2 /* TransactionOperation.h in Headers */,
</span><ins>+ 07C046C21E425022007201E7 /* AudioSampleDataSource.h in Headers */,
</ins><span class="cx"> 49E911C40EF86D47009D0CAF /* TransformationMatrix.h in Headers */,
</span><span class="cx"> FB484F4D171F821E00040755 /* TransformFunctions.h in Headers */,
</span><span class="cx"> 49E911CE0EF86D47009D0CAF /* TransformOperation.h in Headers */,
</span><span class="lines">@@ -29982,7 +29969,6 @@
</span><span class="cx"> 51F798EF1BE880E7008AE491 /* IDBIndexInfo.cpp in Sources */,
</span><span class="cx"> 51E269361DD3BD97006B6A58 /* IDBIterateCursorData.cpp in Sources */,
</span><span class="cx"> 5185FC941BB4C4E80012898F /* IDBKey.cpp in Sources */,
</span><del>- 073B87661E4385AC0071C0EC /* AudioSampleBufferList.cpp in Sources */,
</del><span class="cx"> 5185FC961BB4C4E80012898F /* IDBKeyData.cpp in Sources */,
</span><span class="cx"> 5185FC981BB4C4E80012898F /* IDBKeyPath.cpp in Sources */,
</span><span class="cx"> 5185FC9A1BB4C4E80012898F /* IDBKeyRange.cpp in Sources */,
</span><span class="lines">@@ -29991,7 +29977,6 @@
</span><span class="cx"> 5160712E1BD8307800DBC4F2 /* IDBObjectStoreInfo.cpp in Sources */,
</span><span class="cx"> 5185FCA31BB4C4E80012898F /* IDBOpenDBRequest.cpp in Sources */,
</span><span class="cx"> 5185FCA81BB4C4E80012898F /* IDBRequest.cpp in Sources */,
</span><del>- 073B87681E4385AC0071C0EC /* AudioSampleDataSource.cpp in Sources */,
</del><span class="cx"> 514129981C6976900059E714 /* IDBRequestCompletionEvent.cpp in Sources */,
</span><span class="cx"> 510A58F91BACC7F200C19282 /* IDBRequestData.cpp in Sources */,
</span><span class="cx"> 5145B1091BC48E2E00E86219 /* IDBResourceIdentifier.cpp in Sources */,
</span><span class="lines">@@ -30508,6 +30493,7 @@
</span><span class="cx"> E1284BB210449FFA00EAEB52 /* JSPageTransitionEvent.cpp in Sources */,
</span><span class="cx"> FDA15EB112B03EE1003A583A /* JSPannerNode.cpp in Sources */,
</span><span class="cx"> E51A81DF17298D7700BFCA61 /* JSPerformance.cpp in Sources */,
</span><ins>+ 07C046C11E425022007201E7 /* AudioSampleDataSource.cpp in Sources */,
</ins><span class="cx"> CB38FD511CCF938900592A3F /* JSPerformanceEntry.cpp in Sources */,
</span><span class="cx"> CB38FD571CD21E2A00592A3F /* JSPerformanceEntryCustom.cpp in Sources */,
</span><span class="cx"> A58C59D01E382EAC0047859C /* JSPerformanceMark.cpp in Sources */,
</span></span></pre></div>
<a id="trunkSourceWebCoreplatformaudiomacAudioSampleBufferListcpp"></a>
<div class="delfile"><h4>Deleted: trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.cpp (211609 => 211610)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.cpp 2017-02-03 00:26:00 UTC (rev 211609)
+++ trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.cpp 2017-02-03 00:35:30 UTC (rev 211610)
</span><span class="lines">@@ -1,340 +0,0 @@
</span><del>-/*
- * 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. ``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
- * 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 "AudioSampleBufferList.h"
-
-#if ENABLE(MEDIA_STREAM)
-
-#include "Logging.h"
-#include "VectorMath.h"
-#include <Accelerate/Accelerate.h>
-#include <AudioToolbox/AudioConverter.h>
-
-namespace WebCore {
-
-using namespace VectorMath;
-
-Ref<AudioSampleBufferList> AudioSampleBufferList::create(const CAAudioStreamDescription& format, size_t maximumSampleCount)
-{
- return adoptRef(*new AudioSampleBufferList(format, maximumSampleCount));
-}
-
-AudioSampleBufferList::AudioSampleBufferList(const CAAudioStreamDescription& format, size_t maximumSampleCount)
-{
- 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();
-}
-
-AudioSampleBufferList::~AudioSampleBufferList()
-{
- m_internalFormat = nullptr;
- m_bufferList = nullptr;
-}
-
-void AudioSampleBufferList::setSampleCount(size_t count)
-{
- ASSERT(count <= m_sampleCapacity);
- if (count <= m_sampleCapacity)
- m_sampleCount = count;
-}
-
-void AudioSampleBufferList::applyGain(AudioBufferList& bufferList, float gain, AudioStreamDescription::PCMFormat format)
-{
- for (uint32_t i = 0; i < bufferList.mNumberBuffers; ++i) {
- switch (format) {
- case AudioStreamDescription::Int16: {
- int16_t* buffer = static_cast<int16_t*>(bufferList.mBuffers[i].mData);
- int frameCount = bufferList.mBuffers[i].mDataByteSize / sizeof(int16_t);
- for (int i = 0; i < frameCount; i++)
- buffer[i] *= gain;
- break;
- }
- case AudioStreamDescription::Int32: {
- int32_t* buffer = static_cast<int32_t*>(bufferList.mBuffers[i].mData);
- int frameCount = bufferList.mBuffers[i].mDataByteSize / sizeof(int32_t);
- for (int i = 0; i < frameCount; i++)
- buffer[i] *= gain;
- break;
- break;
- }
- case AudioStreamDescription::Float32: {
- float* buffer = static_cast<float*>(bufferList.mBuffers[i].mData);
- vDSP_vsmul(buffer, 1, &gain, buffer, 1, bufferList.mBuffers[i].mDataByteSize / sizeof(float));
- break;
- }
- case AudioStreamDescription::Float64: {
- double* buffer = static_cast<double*>(bufferList.mBuffers[i].mData);
- double gainAsDouble = gain;
- vDSP_vsmulD(buffer, 1, &gainAsDouble, buffer, 1, bufferList.mBuffers[i].mDataByteSize / sizeof(double));
- break;
- }
- case AudioStreamDescription::None:
- ASSERT_NOT_REACHED();
- break;
- }
- }
-}
-
-void AudioSampleBufferList::applyGain(float gain)
-{
- applyGain(*m_bufferList, gain, m_internalFormat->format());
-}
-
-OSStatus AudioSampleBufferList::mixFrom(const AudioSampleBufferList& source, size_t frameCount)
-{
- ASSERT(source.streamDescription() == streamDescription());
-
- if (source.streamDescription() != streamDescription())
- return kAudio_ParamError;
-
- if (frameCount > source.sampleCount())
- frameCount = source.sampleCount();
-
- if (frameCount > m_sampleCapacity)
- return kAudio_ParamError;
-
- m_sampleCount = frameCount;
-
- AudioBufferList& sourceBuffer = source.bufferList();
- for (uint32_t i = 0; i < m_bufferList->mNumberBuffers; 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);
- 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);
- 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);
- 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);
- break;
- }
- case AudioStreamDescription::None:
- ASSERT_NOT_REACHED();
- break;
- }
- }
-
- return 0;
-}
-
-OSStatus AudioSampleBufferList::copyFrom(const AudioSampleBufferList& source, size_t frameCount)
-{
- ASSERT(source.streamDescription() == streamDescription());
-
- if (source.streamDescription() != streamDescription())
- return kAudio_ParamError;
-
- if (frameCount > source.sampleCount())
- frameCount = source.sampleCount();
-
- if (frameCount > m_sampleCapacity)
- return kAudio_ParamError;
-
- 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);
- memcpy(destination, sourceData, frameCount * m_internalFormat->bytesPerPacket());
- }
-
- return 0;
-}
-
-OSStatus AudioSampleBufferList::copyTo(AudioBufferList& buffer, size_t frameCount)
-{
- if (frameCount > m_sampleCount)
- return kAudio_ParamError;
- if (buffer.mNumberBuffers > m_bufferList->mNumberBuffers)
- 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* destination = static_cast<uint8_t*>(buffer.mBuffers[i].mData);
- memcpy(destination, sourceData, frameCount * m_internalFormat->bytesPerPacket());
- }
-
- return 0;
-}
-
-void AudioSampleBufferList::reset()
-{
- m_sampleCount = 0;
- m_timestamp = 0;
- m_hostTime = -1;
-
- uint8_t* data = reinterpret_cast<uint8_t*>(m_bufferList.get()) + 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 = data + m_maxBufferSizePerChannel;
- }
-}
-
-void AudioSampleBufferList::zero()
-{
- zeroABL(*m_bufferList, m_internalFormat->bytesPerPacket() * m_sampleCapacity);
-}
-
-void AudioSampleBufferList::zeroABL(AudioBufferList& buffer, size_t byteCount)
-{
- for (uint32_t i = 0; i < buffer.mNumberBuffers; ++i)
- memset(buffer.mBuffers[i].mData, 0, byteCount);
-}
-
-OSStatus AudioSampleBufferList::convertInput(UInt32* ioNumberDataPackets, AudioBufferList* ioData)
-{
- if (!ioNumberDataPackets || !ioData || !m_converterInputBuffer) {
- LOG_ERROR("AudioSampleBufferList::reconfigureInput(%p) invalid input to AudioConverterInput", this);
- return kAudioConverterErr_UnspecifiedError;
- }
-
- size_t packetCount = m_converterInputBuffer->mBuffers[0].mDataByteSize / m_converterInputBytesPerPacket;
- if (*ioNumberDataPackets > m_sampleCapacity) {
- LOG_ERROR("AudioSampleBufferList::convertInput(%p) not enough internal storage: needed = %u, available = %lu", this, *ioNumberDataPackets, m_sampleCapacity);
- return kAudioConverterErr_InvalidInputSize;
- }
-
- *ioNumberDataPackets = static_cast<UInt32>(packetCount);
- for (uint32_t i = 0; i < ioData->mNumberBuffers; ++i) {
- ioData->mBuffers[i].mData = m_converterInputBuffer->mBuffers[i].mData;
- ioData->mBuffers[i].mDataByteSize = m_converterInputBuffer->mBuffers[i].mDataByteSize;
- }
-
- return 0;
-}
-
-OSStatus AudioSampleBufferList::audioConverterCallback(AudioConverterRef, UInt32* ioNumberDataPackets, AudioBufferList* ioData, AudioStreamPacketDescription**, void* inRefCon)
-{
- return static_cast<AudioSampleBufferList*>(inRefCon)->convertInput(ioNumberDataPackets, ioData);
-}
-
-OSStatus AudioSampleBufferList::copyFrom(AudioBufferList& source, AudioConverterRef converter)
-{
- reset();
-
- AudioStreamBasicDescription inputFormat;
- UInt32 propertyDataSize = sizeof(inputFormat);
- AudioConverterGetProperty(converter, kAudioConverterCurrentInputStreamDescription, &propertyDataSize, &inputFormat);
- 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);
- }
-#endif
-
- UInt32 samplesConverted = static_cast<UInt32>(m_sampleCapacity);
- OSStatus err = AudioConverterFillComplexBuffer(converter, audioConverterCallback, this, &samplesConverted, outputData, nullptr);
- if (err) {
- LOG_ERROR("AudioSampleBufferList::copyFrom(%p) AudioConverterFillComplexBuffer returned error %d (%.4s)", this, err, (char*)&err);
- m_sampleCount = std::min(m_sampleCapacity, static_cast<size_t>(samplesConverted));
- zero();
- return err;
- }
-
- m_sampleCount = samplesConverted;
- return 0;
-}
-
-OSStatus AudioSampleBufferList::copyFrom(AudioSampleBufferList& source, AudioConverterRef converter)
-{
- return copyFrom(source.bufferList(), converter);
-}
-
-OSStatus AudioSampleBufferList::copyFrom(CARingBuffer& ringBuffer, size_t sampleCount, uint64_t startFrame, CARingBuffer::FetchMode mode)
-{
- reset();
- if (ringBuffer.fetch(&bufferList(), sampleCount, startFrame, mode) != CARingBuffer::Ok)
- return kAudio_ParamError;
-
- m_sampleCount = sampleCount;
- 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)
</del></span></pre></div>
<a id="trunkSourceWebCoreplatformaudiomacAudioSampleBufferListh"></a>
<div class="delfile"><h4>Deleted: trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.h (211609 => 211610)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.h 2017-02-03 00:26:00 UTC (rev 211609)
+++ trunk/Source/WebCore/platform/audio/mac/AudioSampleBufferList.h 2017-02-03 00:35:30 UTC (rev 211610)
</span><span class="lines">@@ -1,106 +0,0 @@
</span><del>-/*
- * 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. ``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
- * 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 ENABLE(MEDIA_STREAM)
-
-#include "CARingBuffer.h"
-#include <CoreAudio/CoreAudioTypes.h>
-#include <wtf/Lock.h>
-#include <wtf/RefCounted.h>
-#include <wtf/RefPtr.h>
-
-typedef struct AudioStreamBasicDescription AudioStreamBasicDescription;
-typedef struct OpaqueAudioConverter* AudioConverterRef;
-
-namespace WebCore {
-
-class AudioSampleBufferList : public RefCounted<AudioSampleBufferList> {
-public:
- static Ref<AudioSampleBufferList> create(const CAAudioStreamDescription&, size_t);
-
- ~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);
- void applyGain(float);
-
- OSStatus copyFrom(const AudioSampleBufferList&, size_t count = SIZE_MAX);
- OSStatus copyFrom(AudioBufferList&, AudioConverterRef);
- OSStatus copyFrom(AudioSampleBufferList&, AudioConverterRef);
- OSStatus copyFrom(CARingBuffer&, size_t frameCount, uint64_t startFrame, CARingBuffer::FetchMode);
-
- OSStatus mixFrom(const AudioSampleBufferList&, size_t count = SIZE_MAX);
-
- OSStatus copyTo(AudioBufferList&, size_t count = SIZE_MAX);
-
- const AudioStreamBasicDescription& streamDescription() const { return m_internalFormat->streamDescription(); }
- AudioBufferList& bufferList() const { return *m_bufferList.get(); }
-
- uint32_t sampleCapacity() const { return m_sampleCapacity; }
- uint32_t sampleCount() const { return m_sampleCount; }
- void setSampleCount(size_t);
-
- uint64_t timestamp() const { return m_timestamp; }
- double hostTime() const { return m_hostTime; }
- void setTimes(uint64_t time, double hostTime) { m_timestamp = time; m_hostTime = hostTime; }
-
- void reset();
-
- static void zeroABL(AudioBufferList&, size_t);
- void zero();
-
-protected:
- AudioSampleBufferList(const CAAudioStreamDescription&, size_t);
-
- static OSStatus audioConverterCallback(AudioConverterRef, UInt32*, AudioBufferList*, AudioStreamPacketDescription**, void*);
- OSStatus convertInput(UInt32*, AudioBufferList*);
-
- std::unique_ptr<CAAudioStreamDescription> m_internalFormat;
-
- AudioSampleBufferList* m_converterInputBuffer2 { nullptr };
- AudioBufferList* m_converterInputBuffer { nullptr };
- uint32_t m_converterInputBytesPerPacket { 0 };
-
- uint64_t m_timestamp { 0 };
- double m_hostTime { -1 };
- size_t m_sampleCount { 0 };
- size_t m_sampleCapacity { 0 };
- size_t m_maxBufferSizePerChannel { 0 };
- size_t m_bufferListBaseSize { 0 };
- std::unique_ptr<AudioBufferList> m_bufferList;
-};
-
-inline size_t AudioSampleBufferList::audioBufferListSizeForStream(const CAAudioStreamDescription& description)
-{
- return offsetof(AudioBufferList, mBuffers) + (sizeof(AudioBuffer) * std::max<uint32_t>(1, description.numberOfChannelStreams()));
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(MEDIA_STREAM)
</del></span></pre></div>
<a id="trunkSourceWebCoreplatformaudiomacAudioSampleDataSourcecpp"></a>
<div class="delfile"><h4>Deleted: trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp (211609 => 211610)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp 2017-02-03 00:26:00 UTC (rev 211609)
+++ trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp 2017-02-03 00:35:30 UTC (rev 211610)
</span><span class="lines">@@ -1,338 +0,0 @@
</span><del>-/*
- * 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. ``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
- * 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 "AudioSampleDataSource.h"
-
-#if ENABLE(MEDIA_STREAM)
-
-#include "CAAudioStreamDescription.h"
-#include "CARingBuffer.h"
-#include "Logging.h"
-#include "MediaTimeAVFoundation.h"
-#include <AudioToolbox/AudioConverter.h>
-#include <mach/mach.h>
-#include <mach/mach_time.h>
-#include <mutex>
-#include <syslog.h>
-#include <wtf/CurrentTime.h>
-#include <wtf/StringPrintStream.h>
-
-#include "CoreMediaSoftLink.h"
-
-namespace WebCore {
-
-using namespace JSC;
-
-Ref<AudioSampleDataSource> AudioSampleDataSource::create(size_t maximumSampleCount)
-{
- return adoptRef(*new AudioSampleDataSource(maximumSampleCount));
-}
-
-AudioSampleDataSource::AudioSampleDataSource(size_t maximumSampleCount)
- : m_inputSampleOffset(MediaTime::invalidTime())
- , m_maximumSampleCount(maximumSampleCount)
-{
-}
-
-AudioSampleDataSource::~AudioSampleDataSource()
-{
- m_inputDescription = nullptr;
- m_outputDescription = nullptr;
- m_ringBuffer = nullptr;
- if (m_converter) {
- AudioConverterDispose(m_converter);
- m_converter = nullptr;
- }
-}
-
-void AudioSampleDataSource::setPaused(bool paused)
-{
- std::lock_guard<Lock> lock(m_lock);
-
- if (paused == m_paused)
- return;
-
- m_transitioningFromPaused = m_paused;
- m_paused = paused;
-}
-
-OSStatus AudioSampleDataSource::setupConverter()
-{
- ASSERT(m_inputDescription && m_outputDescription);
-
- if (m_converter) {
- AudioConverterDispose(m_converter);
- m_converter = nullptr;
- }
-
- if (*m_inputDescription == *m_outputDescription)
- return 0;
-
- OSStatus err = AudioConverterNew(&m_inputDescription->streamDescription(), &m_outputDescription->streamDescription(), &m_converter);
- if (err)
- LOG_ERROR("AudioSampleDataSource::setupConverter(%p) - AudioConverterNew returned error %d (%.4s)", this, (int)err, (char*)&err);
-
- return err;
-
-}
-
-OSStatus AudioSampleDataSource::setInputFormat(const CAAudioStreamDescription& format)
-{
- ASSERT(format.sampleRate() >= 0);
-
- m_inputDescription = std::make_unique<CAAudioStreamDescription>(format);
- if (m_outputDescription)
- return setupConverter();
-
- return 0;
-}
-
-OSStatus AudioSampleDataSource::setOutputFormat(const CAAudioStreamDescription& format)
-{
- ASSERT(m_inputDescription);
- ASSERT(format.sampleRate() >= 0);
-
- m_outputDescription = std::make_unique<CAAudioStreamDescription>(format);
- if (!m_ringBuffer)
- m_ringBuffer = std::make_unique<CARingBuffer>();
-
- m_ringBuffer->allocate(format, static_cast<size_t>(m_maximumSampleCount));
- m_scratchBuffer = AudioSampleBufferList::create(m_outputDescription->streamDescription(), m_maximumSampleCount);
-
- return setupConverter();
-}
-
-MediaTime AudioSampleDataSource::hostTime() const
-{
- // Based on listing #2 from Apple Technical Q&A QA1398, modified to be thread-safe.
- static double frequency;
- static mach_timebase_info_data_t timebaseInfo;
- static std::once_flag initializeTimerOnceFlag;
- std::call_once(initializeTimerOnceFlag, [] {
- kern_return_t kr = mach_timebase_info(&timebaseInfo);
- frequency = 1e-9 * static_cast<double>(timebaseInfo.numer) / static_cast<double>(timebaseInfo.denom);
- ASSERT_UNUSED(kr, kr == KERN_SUCCESS);
- ASSERT(timebaseInfo.denom);
- });
-
- return MediaTime::createWithDouble(mach_absolute_time() * frequency);
-}
-
-void AudioSampleDataSource::pushSamplesInternal(AudioBufferList& bufferList, const MediaTime& presentationTime, size_t sampleCount)
-{
- ASSERT(m_lock.isHeld());
-
- AudioBufferList* sampleBufferList;
- if (m_converter) {
- m_scratchBuffer->reset();
- OSStatus err = m_scratchBuffer->copyFrom(bufferList, m_converter);
- if (err)
- return;
-
- sampleBufferList = &m_scratchBuffer->bufferList();
- } else
- sampleBufferList = &bufferList;
-
- MediaTime sampleTime = presentationTime;
- if (m_inputSampleOffset == MediaTime::invalidTime()) {
- m_inputSampleOffset = MediaTime(1 - sampleTime.timeValue(), sampleTime.timeScale());
- if (m_inputSampleOffset.timeScale() != sampleTime.timeScale()) {
- // FIXME: It should be possible to do this without calling CMTimeConvertScale.
- m_inputSampleOffset = toMediaTime(CMTimeConvertScale(toCMTime(m_inputSampleOffset), sampleTime.timeScale(), kCMTimeRoundingMethod_Default));
- }
- LOG(MediaCaptureSamples, "@@ pushSamples: input sample offset is %lld, m_maximumSampleCount = %zu", m_inputSampleOffset.timeValue(), m_maximumSampleCount);
- }
- sampleTime += m_inputSampleOffset;
-
-#if !LOG_DISABLED
- uint64_t startFrame1 = 0;
- uint64_t endFrame1 = 0;
- m_ringBuffer->getCurrentFrameBounds(startFrame1, endFrame1);
-#endif
-
- m_ringBuffer->store(sampleBufferList, sampleCount, sampleTime.timeValue());
- m_timeStamp = sampleTime.timeValue();
-
- LOG(MediaCaptureSamples, "@@ pushSamples: added %ld samples for time = %s (was %s), mach time = %lld", sampleCount, toString(sampleTime).utf8().data(), toString(presentationTime).utf8().data(), mach_absolute_time());
-
-#if !LOG_DISABLED
- uint64_t startFrame2 = 0;
- uint64_t endFrame2 = 0;
- m_ringBuffer->getCurrentFrameBounds(startFrame2, endFrame2);
- LOG(MediaCaptureSamples, "@@ pushSamples: buffered range was [%lld .. %lld], is [%lld .. %lld]", startFrame1, endFrame1, startFrame2, endFrame2);
-#endif
-}
-
-void AudioSampleDataSource::pushSamples(const AudioStreamBasicDescription& sampleDescription, CMSampleBufferRef sampleBuffer)
-{
- std::lock_guard<Lock> lock(m_lock);
-
- 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));
-}
-
-void AudioSampleDataSource::pushSamples(const AudioStreamBasicDescription& sampleDescription, const MediaTime& sampleTime, void* 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);
-}
-
-bool AudioSampleDataSource::pullSamplesInternal(AudioBufferList& buffer, size_t& sampleCount, uint64_t timeStamp, double /*hostTime*/, PullMode mode)
-{
- ASSERT(m_lock.isHeld());
-
- ASSERT(buffer.mNumberBuffers == m_ringBuffer->channelCount());
- if (buffer.mNumberBuffers != m_ringBuffer->channelCount()) {
- AudioSampleBufferList::zeroABL(buffer, sampleCount);
- sampleCount = 0;
- return false;
- }
-
- if (!m_ringBuffer || m_muted || m_inputSampleOffset == MediaTime::invalidTime()) {
- AudioSampleBufferList::zeroABL(buffer, sampleCount);
- sampleCount = 0;
- return false;
- }
-
- uint64_t startFrame = 0;
- uint64_t endFrame = 0;
- m_ringBuffer->getCurrentFrameBounds(startFrame, endFrame);
-
- if (m_transitioningFromPaused) {
- uint64_t buffered = endFrame - m_timeStamp;
- if (buffered < sampleCount * 2) {
- AudioSampleBufferList::zeroABL(buffer, sampleCount);
- sampleCount = 0;
- return false;
- }
-
- const double twentyMS = .02;
- const double tenMS = .01;
- const double fiveMS = .005;
- double sampleRate = m_outputDescription->sampleRate();
- if (buffered > sampleRate * twentyMS)
- m_outputSampleOffset = m_timeStamp - sampleRate * twentyMS;
- else if (buffered > sampleRate * tenMS)
- m_outputSampleOffset = m_timeStamp - sampleRate * tenMS;
- else if (buffered > sampleRate * fiveMS)
- m_outputSampleOffset = m_timeStamp - sampleRate * fiveMS;
- else
- m_outputSampleOffset = m_timeStamp;
-
- m_transitioningFromPaused = false;
- }
-
- timeStamp += m_outputSampleOffset;
-
- LOG(MediaCaptureSamples, "** pullSamples: asking for %ld samples at time = %lld (was %lld)", sampleCount, timeStamp, timeStamp - m_outputSampleOffset);
-
- int64_t framesAvailable = sampleCount;
- if (timeStamp < startFrame || timeStamp + sampleCount > endFrame) {
- if (timeStamp + sampleCount < startFrame || timeStamp > endFrame)
- framesAvailable = 0;
- else if (timeStamp < startFrame)
- framesAvailable = timeStamp + sampleCount - startFrame;
- else
- framesAvailable = timeStamp + sampleCount - endFrame;
-
- LOG(MediaCaptureSamples, "** pullSamplesInternal: sample %lld is not completely in range [%lld .. %lld], returning %lld frames", timeStamp, startFrame, endFrame, framesAvailable);
-
- if (!framesAvailable) {
- AudioSampleBufferList::zeroABL(buffer, sampleCount);
- return false;
- }
- }
-
- if (m_volume >= .95) {
- m_ringBuffer->fetch(&buffer, sampleCount, timeStamp, mode == Copy ? CARingBuffer::Copy : CARingBuffer::Mix);
- return true;
- }
-
- if (m_scratchBuffer->copyFrom(*m_ringBuffer.get(), sampleCount, timeStamp, mode == Copy ? CARingBuffer::Copy : CARingBuffer::Mix)) {
- AudioSampleBufferList::zeroABL(buffer, sampleCount);
- return false;
- }
-
- m_scratchBuffer->applyGain(m_volume);
- if (m_scratchBuffer->copyTo(buffer, sampleCount))
- AudioSampleBufferList::zeroABL(buffer, sampleCount);
-
- return true;
-}
-
-bool AudioSampleDataSource::pullSamples(AudioBufferList& buffer, size_t sampleCount, uint64_t timeStamp, double hostTime, PullMode mode)
-{
- std::unique_lock<Lock> lock(m_lock, std::try_to_lock);
- if (!lock.owns_lock() || !m_ringBuffer) {
- AudioSampleBufferList::zeroABL(buffer, sampleCount);
- return false;
- }
-
- return pullSamplesInternal(buffer, sampleCount, timeStamp, hostTime, mode);
-}
-
-bool AudioSampleDataSource::pullSamples(AudioSampleBufferList& buffer, size_t sampleCount, uint64_t timeStamp, double hostTime, PullMode mode)
-{
- std::unique_lock<Lock> lock(m_lock, std::try_to_lock);
- if (!lock.owns_lock() || !m_ringBuffer) {
- buffer.zero();
- return false;
- }
-
- if (!pullSamplesInternal(buffer.bufferList(), sampleCount, timeStamp, hostTime, mode))
- return false;
-
- buffer.setTimes(timeStamp, hostTime);
- buffer.setSampleCount(sampleCount);
-
- return true;
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(MEDIA_STREAM)
</del></span></pre></div>
<a id="trunkSourceWebCoreplatformaudiomacAudioSampleDataSourceh"></a>
<div class="delfile"><h4>Deleted: trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.h (211609 => 211610)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.h 2017-02-03 00:26:00 UTC (rev 211609)
+++ trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.h 2017-02-03 00:35:30 UTC (rev 211610)
</span><span class="lines">@@ -1,104 +0,0 @@
</span><del>-/*
- * 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. ``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
- * 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 ENABLE(MEDIA_STREAM)
-
-#include "AudioSampleBufferList.h"
-#include <CoreAudio/CoreAudioTypes.h>
-#include <wtf/Lock.h>
-#include <wtf/MediaTime.h>
-#include <wtf/RefCounted.h>
-#include <wtf/RefPtr.h>
-#include <wtf/text/WTFString.h>
-
-typedef const struct opaqueCMFormatDescription *CMFormatDescriptionRef;
-typedef struct opaqueCMSampleBuffer *CMSampleBufferRef;
-
-namespace WebCore {
-
-class CAAudioStreamDescription;
-class CARingBuffer;
-
-class AudioSampleDataSource : public RefCounted<AudioSampleDataSource> {
-public:
- static Ref<AudioSampleDataSource> create(size_t);
-
- ~AudioSampleDataSource();
-
- OSStatus setInputFormat(const CAAudioStreamDescription&);
- OSStatus setOutputFormat(const CAAudioStreamDescription&);
-
- void pushSamples(const AudioStreamBasicDescription&, const MediaTime&, void*, size_t);
- void pushSamples(const AudioStreamBasicDescription&, CMSampleBufferRef);
-
- enum PullMode { Copy, Mix };
- bool pullSamples(AudioSampleBufferList&, size_t, uint64_t, double, PullMode);
- bool pullSamples(AudioBufferList&, size_t, uint64_t, double, PullMode);
-
- void setPaused(bool);
-
- void setVolume(float volume) { m_volume = volume; }
- float volume() const { return m_volume; }
-
- void setMuted(bool muted) { m_muted = muted; }
- bool muted() const { return m_muted; }
-
-protected:
- AudioSampleDataSource(size_t);
-
- OSStatus setupConverter();
- bool pullSamplesInternal(AudioBufferList&, size_t&, uint64_t, double, PullMode);
-
- void pushSamplesInternal(AudioBufferList&, const MediaTime&, size_t frameCount);
-
- std::unique_ptr<CAAudioStreamDescription> m_inputDescription;
- std::unique_ptr<CAAudioStreamDescription> m_outputDescription;
-
- MediaTime hostTime() const;
-
- uint64_t m_timeStamp { 0 };
- double m_hostTime { -1 };
-
- MediaTime m_inputSampleOffset;
- uint64_t m_outputSampleOffset { 0 };
-
- AudioConverterRef m_converter;
- RefPtr<AudioSampleBufferList> m_scratchBuffer;
-
- std::unique_ptr<CARingBuffer> m_ringBuffer;
- size_t m_maximumSampleCount { 0 };
-
- Lock m_lock;
- float m_volume { 1.0 };
- bool m_muted { false };
- bool m_paused { true };
- bool m_transitioningFromPaused { true };
-};
-
-} // namespace WebCore
-
-#endif // ENABLE(MEDIA_STREAM)
</del></span></pre>
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