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<title>[212727] trunk/Source/WebCore</title>
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<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/212727">212727</a></dd>
<dt>Author</dt> <dd>jer.noble@apple.com</dd>
<dt>Date</dt> <dd>2017-02-21 10:45:10 -0800 (Tue, 21 Feb 2017)</dd>
</dl>
<h3>Log Message</h3>
<pre>Make logging in high-priority audio threads less expensive
https://bugs.webkit.org/show_bug.cgi?id=168639
Reviewed by Jon Lee.
Logging from inside a high-priority audio thread will make a number of calls to malloc, block, and
therefore cause audio glitches. Make this logging less expensive by dispatching to the main thread
before creating and outputting the log string.
* WebCore.xcodeproj/project.pbxproj:
* platform/audio/mac/AudioSampleDataSource.mm: Renamed from Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp.
(WebCore::AudioSampleDataSource::pushSamplesInternal):
(WebCore::AudioSampleDataSource::pullSamplesInternal):</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>Added Paths</h3>
<ul>
<li><a href="#trunkSourceWebCoreplatformaudiomacAudioSampleDataSourcemm">trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.mm</a></li>
</ul>
<h3>Removed Paths</h3>
<ul>
<li><a href="#trunkSourceWebCoreplatformaudiomacAudioSampleDataSourcecpp">trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkSourceWebCoreChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/ChangeLog (212726 => 212727)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/ChangeLog 2017-02-21 18:36:00 UTC (rev 212726)
+++ trunk/Source/WebCore/ChangeLog 2017-02-21 18:45:10 UTC (rev 212727)
</span><span class="lines">@@ -1,5 +1,21 @@
</span><span class="cx"> 2017-02-21 Jer Noble <jer.noble@apple.com>
</span><span class="cx">
</span><ins>+ Make logging in high-priority audio threads less expensive
+ https://bugs.webkit.org/show_bug.cgi?id=168639
+
+ Reviewed by Jon Lee.
+
+ Logging from inside a high-priority audio thread will make a number of calls to malloc, block, and
+ therefore cause audio glitches. Make this logging less expensive by dispatching to the main thread
+ before creating and outputting the log string.
+
+ * WebCore.xcodeproj/project.pbxproj:
+ * platform/audio/mac/AudioSampleDataSource.mm: Renamed from Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp.
+ (WebCore::AudioSampleDataSource::pushSamplesInternal):
+ (WebCore::AudioSampleDataSource::pullSamplesInternal):
+
+2017-02-21 Jer Noble <jer.noble@apple.com>
+
</ins><span class="cx"> Give the Mock audio input a "hum" to make drop-outs more detectable
</span><span class="cx"> https://bugs.webkit.org/show_bug.cgi?id=168641
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoreWebCorexcodeprojprojectpbxproj"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj (212726 => 212727)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2017-02-21 18:36:00 UTC (rev 212726)
+++ trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2017-02-21 18:45:10 UTC (rev 212727)
</span><span class="lines">@@ -144,7 +144,7 @@
</span><span class="cx"> 07394ECA1BAB2CD700BE99CD /* MediaDevicesRequest.h in Headers */ = {isa = PBXBuildFile; fileRef = 07394EC91BAB2CD700BE99CD /* MediaDevicesRequest.h */; settings = {ATTRIBUTES = (Private, ); }; };
</span><span class="cx"> 073B87661E4385AC0071C0EC /* AudioSampleBufferList.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 073B87621E43859D0071C0EC /* AudioSampleBufferList.cpp */; };
</span><span class="cx"> 073B87671E4385AC0071C0EC /* AudioSampleBufferList.h in Headers */ = {isa = PBXBuildFile; fileRef = 073B87631E43859D0071C0EC /* AudioSampleBufferList.h */; };
</span><del>- 073B87681E4385AC0071C0EC /* AudioSampleDataSource.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 073B87641E43859D0071C0EC /* AudioSampleDataSource.cpp */; };
</del><ins>+ 073B87681E4385AC0071C0EC /* AudioSampleDataSource.mm in Sources */ = {isa = PBXBuildFile; fileRef = 073B87641E43859D0071C0EC /* AudioSampleDataSource.mm */; };
</ins><span class="cx"> 073B87691E4385AC0071C0EC /* AudioSampleDataSource.h in Headers */ = {isa = PBXBuildFile; fileRef = 073B87651E43859D0071C0EC /* AudioSampleDataSource.h */; };
</span><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="lines">@@ -7270,7 +7270,7 @@
</span><span class="cx"> 073B87581E40DCFD0071C0EC /* CAAudioStreamDescription.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CAAudioStreamDescription.h; sourceTree = "<group>"; };
</span><span class="cx"> 073B87621E43859D0071C0EC /* AudioSampleBufferList.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = AudioSampleBufferList.cpp; sourceTree = "<group>"; };
</span><span class="cx"> 073B87631E43859D0071C0EC /* AudioSampleBufferList.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = AudioSampleBufferList.h; sourceTree = "<group>"; };
</span><del>- 073B87641E43859D0071C0EC /* AudioSampleDataSource.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = AudioSampleDataSource.cpp; sourceTree = "<group>"; };
</del><ins>+ 073B87641E43859D0071C0EC /* AudioSampleDataSource.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = AudioSampleDataSource.mm; sourceTree = "<group>"; };
</ins><span class="cx"> 073B87651E43859D0071C0EC /* AudioSampleDataSource.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = AudioSampleDataSource.h; sourceTree = "<group>"; };
</span><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="lines">@@ -25107,7 +25107,7 @@
</span><span class="cx"> CD2F4A2618D8A3490063746D /* AudioHardwareListenerMac.h */,
</span><span class="cx"> 073B87621E43859D0071C0EC /* AudioSampleBufferList.cpp */,
</span><span class="cx"> 073B87631E43859D0071C0EC /* AudioSampleBufferList.h */,
</span><del>- 073B87641E43859D0071C0EC /* AudioSampleDataSource.cpp */,
</del><ins>+ 073B87641E43859D0071C0EC /* AudioSampleDataSource.mm */,
</ins><span class="cx"> 073B87651E43859D0071C0EC /* AudioSampleDataSource.h */,
</span><span class="cx"> CD54DE4917469C6D005E5B36 /* AudioSessionMac.cpp */,
</span><span class="cx"> 073B87571E40DCFD0071C0EC /* CAAudioStreamDescription.cpp */,
</span><span class="lines">@@ -29327,7 +29327,7 @@
</span><span class="cx"> FD31608512B026F700C1A359 /* AudioResampler.cpp in Sources */,
</span><span class="cx"> FD31608712B026F700C1A359 /* AudioResamplerKernel.cpp in Sources */,
</span><span class="cx"> 073B87661E4385AC0071C0EC /* AudioSampleBufferList.cpp in Sources */,
</span><del>- 073B87681E4385AC0071C0EC /* AudioSampleDataSource.cpp in Sources */,
</del><ins>+ 073B87681E4385AC0071C0EC /* AudioSampleDataSource.mm in Sources */,
</ins><span class="cx"> FD8C46EB154608E700A5910C /* AudioScheduledSourceNode.cpp in Sources */,
</span><span class="cx"> CDA79824170A258300D45C55 /* AudioSession.cpp in Sources */,
</span><span class="cx"> CDA79827170A279100D45C55 /* AudioSessionIOS.mm in Sources */,
</span></span></pre></div>
<a id="trunkSourceWebCoreplatformaudiomacAudioSampleDataSourcecpp"></a>
<div class="delfile"><h4>Deleted: trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp (212726 => 212727)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp 2017-02-21 18:36:00 UTC (rev 212726)
+++ trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp 2017-02-21 18:45:10 UTC (rev 212727)
</span><span class="lines">@@ -1,355 +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(const AudioBufferList& bufferList, const MediaTime& presentationTime, size_t sampleCount)
-{
- ASSERT(m_lock.isHeld());
-
- const AudioBufferList* sampleBufferList;
- if (m_converter) {
- m_scratchBuffer->reset();
- OSStatus err = m_scratchBuffer->copyFrom(bufferList, m_converter);
- if (err)
- return;
-
- sampleBufferList = m_scratchBuffer->bufferList().list();
- } 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);
-
- WebAudioBufferList list(*m_inputDescription, sampleBuffer);
- pushSamplesInternal(list, toMediaTime(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)), CMSampleBufferGetNumSamples(sampleBuffer));
-}
-
-void AudioSampleDataSource::pushSamples(const MediaTime& sampleTime, const PlatformAudioData& audioData, size_t sampleCount)
-{
- std::unique_lock<Lock> lock(m_lock, std::try_to_lock);
- ASSERT(is<WebAudioBufferList>(audioData));
- pushSamplesInternal(*downcast<WebAudioBufferList>(audioData).list(), sampleTime, sampleCount);
-}
-
-bool AudioSampleDataSource::pullSamplesInternal(AudioBufferList& buffer, size_t& sampleCount, uint64_t timeStamp, double /*hostTime*/, PullMode mode)
-{
- ASSERT(m_lock.isHeld());
- size_t byteCount = sampleCount * m_outputDescription->bytesPerFrame();
-
- ASSERT(buffer.mNumberBuffers == m_ringBuffer->channelCount());
- if (buffer.mNumberBuffers != m_ringBuffer->channelCount()) {
- AudioSampleBufferList::zeroABL(buffer, byteCount);
- sampleCount = 0;
- return false;
- }
-
- if (!m_ringBuffer || m_muted || m_inputSampleOffset == MediaTime::invalidTime()) {
- AudioSampleBufferList::zeroABL(buffer, byteCount);
- 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, byteCount);
- sampleCount = 0;
- return false;
- }
-
- const double twentyMS = .02;
- const double tenMS = .01;
- const double fiveMS = .005;
- double sampleRate = m_outputDescription->sampleRate();
- m_outputSampleOffset = timeStamp + m_timeStamp;
- if (buffered > sampleRate * twentyMS)
- m_outputSampleOffset -= sampleRate * twentyMS;
- else if (buffered > sampleRate * tenMS)
- m_outputSampleOffset -= sampleRate * tenMS;
- else if (buffered > sampleRate * fiveMS)
- m_outputSampleOffset -= sampleRate * fiveMS;
-
- m_transitioningFromPaused = false;
- }
-
- timeStamp += m_outputSampleOffset;
-
- LOG(MediaCaptureSamples, "** pullSamples: asking for %ld samples at time = %lld (was %lld)", sampleCount, timeStamp, timeStamp - m_outputSampleOffset);
-
- uint64_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 < sampleCount) {
- const double twentyMS = .02;
- double sampleRate = m_outputDescription->sampleRate();
- auto delta = static_cast<int64_t>(timeStamp) - endFrame;
- if (delta > 0 && delta < sampleRate * twentyMS)
- m_outputSampleOffset -= delta;
- }
-
- if (!framesAvailable) {
- AudioSampleBufferList::zeroABL(buffer, byteCount);
- 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, byteCount);
-
- return true;
-}
-
-bool AudioSampleDataSource::pullAvalaibleSamplesAsChunks(AudioBufferList& buffer, size_t sampleCountPerChunk, uint64_t timeStamp, Function<void()>&& consumeFilledBuffer)
-{
- std::unique_lock<Lock> lock(m_lock, std::try_to_lock);
- if (!lock.owns_lock() || !m_ringBuffer)
- return false;
-
- ASSERT(buffer.mNumberBuffers == m_ringBuffer->channelCount());
- if (buffer.mNumberBuffers != m_ringBuffer->channelCount())
- return false;
-
- uint64_t startFrame = 0;
- uint64_t endFrame = 0;
- m_ringBuffer->getCurrentFrameBounds(startFrame, endFrame);
- if (timeStamp < startFrame)
- return false;
-
- startFrame = timeStamp;
- while (endFrame - startFrame >= sampleCountPerChunk) {
- if (m_ringBuffer->fetch(&buffer, sampleCountPerChunk, startFrame, CARingBuffer::Copy))
- return false;
- consumeFilledBuffer();
- startFrame += sampleCountPerChunk;
- }
- 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) {
- size_t byteCount = sampleCount * m_outputDescription->bytesPerFrame();
- AudioSampleBufferList::zeroABL(buffer, byteCount);
- 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="trunkSourceWebCoreplatformaudiomacAudioSampleDataSourcemmfromrev212726trunkSourceWebCoreplatformaudiomacAudioSampleDataSourcecpp"></a>
<div class="copfile"><h4>Copied: trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.mm (from rev 212726, trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.cpp) (0 => 212727)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.mm (rev 0)
+++ trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.mm 2017-02-21 18:45:10 UTC (rev 212727)
</span><span class="lines">@@ -0,0 +1,365 @@
</span><ins>+/*
+ * 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(const AudioBufferList& bufferList, const MediaTime& presentationTime, size_t sampleCount)
+{
+ ASSERT(m_lock.isHeld());
+
+ const AudioBufferList* sampleBufferList;
+ if (m_converter) {
+ m_scratchBuffer->reset();
+ OSStatus err = m_scratchBuffer->copyFrom(bufferList, m_converter);
+ if (err)
+ return;
+
+ sampleBufferList = m_scratchBuffer->bufferList().list();
+ } 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();
+
+
+#if !LOG_DISABLED
+ uint64_t startFrame2 = 0;
+ uint64_t endFrame2 = 0;
+ m_ringBuffer->getCurrentFrameBounds(startFrame2, endFrame2);
+ dispatch_async(dispatch_get_main_queue(), [sampleCount, sampleTime, presentationTime, absoluteTime = mach_absolute_time(), startFrame1, endFrame1, startFrame2, endFrame2] {
+ LOG(MediaCaptureSamples, "@@ pushSamples: added %ld samples for time = %s (was %s), mach time = %lld", sampleCount, toString(sampleTime).utf8().data(), toString(presentationTime).utf8().data(), absoluteTime);
+ 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);
+
+ WebAudioBufferList list(*m_inputDescription, sampleBuffer);
+ pushSamplesInternal(list, toMediaTime(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)), CMSampleBufferGetNumSamples(sampleBuffer));
+}
+
+void AudioSampleDataSource::pushSamples(const MediaTime& sampleTime, const PlatformAudioData& audioData, size_t sampleCount)
+{
+ std::unique_lock<Lock> lock(m_lock, std::try_to_lock);
+ ASSERT(is<WebAudioBufferList>(audioData));
+ pushSamplesInternal(*downcast<WebAudioBufferList>(audioData).list(), sampleTime, sampleCount);
+}
+
+bool AudioSampleDataSource::pullSamplesInternal(AudioBufferList& buffer, size_t& sampleCount, uint64_t timeStamp, double /*hostTime*/, PullMode mode)
+{
+ ASSERT(m_lock.isHeld());
+ size_t byteCount = sampleCount * m_outputDescription->bytesPerFrame();
+
+ ASSERT(buffer.mNumberBuffers == m_ringBuffer->channelCount());
+ if (buffer.mNumberBuffers != m_ringBuffer->channelCount()) {
+ AudioSampleBufferList::zeroABL(buffer, byteCount);
+ sampleCount = 0;
+ return false;
+ }
+
+ if (!m_ringBuffer || m_muted || m_inputSampleOffset == MediaTime::invalidTime()) {
+ AudioSampleBufferList::zeroABL(buffer, byteCount);
+ 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, byteCount);
+ sampleCount = 0;
+ return false;
+ }
+
+ const double twentyMS = .02;
+ const double tenMS = .01;
+ const double fiveMS = .005;
+ double sampleRate = m_outputDescription->sampleRate();
+ m_outputSampleOffset = timeStamp + m_timeStamp;
+ if (buffered > sampleRate * twentyMS)
+ m_outputSampleOffset -= sampleRate * twentyMS;
+ else if (buffered > sampleRate * tenMS)
+ m_outputSampleOffset -= sampleRate * tenMS;
+ else if (buffered > sampleRate * fiveMS)
+ m_outputSampleOffset -= sampleRate * fiveMS;
+
+ m_transitioningFromPaused = false;
+ }
+
+ timeStamp += m_outputSampleOffset;
+
+#if !LOG_DISABLED
+ dispatch_async(dispatch_get_main_queue(), [sampleCount, timeStamp, sampleOffset = m_outputSampleOffset] {
+ LOG(MediaCaptureSamples, "** pullSamples: asking for %ld samples at time = %lld (was %lld)", sampleCount, timeStamp, timeStamp - sampleOffset);
+ });
+#endif
+
+ uint64_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;
+
+#if !LOG_DISABLED
+ dispatch_async(dispatch_get_main_queue(), [timeStamp, startFrame, endFrame, framesAvailable] {
+ LOG(MediaCaptureSamples, "** pullSamplesInternal: sample %lld is not completely in range [%lld .. %lld], returning %lld frames", timeStamp, startFrame, endFrame, framesAvailable);
+ });
+#endif
+
+ if (framesAvailable < sampleCount) {
+ const double twentyMS = .02;
+ double sampleRate = m_outputDescription->sampleRate();
+ auto delta = static_cast<int64_t>(timeStamp) - endFrame;
+ if (delta > 0 && delta < sampleRate * twentyMS)
+ m_outputSampleOffset -= delta;
+ }
+
+ if (!framesAvailable) {
+ AudioSampleBufferList::zeroABL(buffer, byteCount);
+ 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, byteCount);
+
+ return true;
+}
+
+bool AudioSampleDataSource::pullAvalaibleSamplesAsChunks(AudioBufferList& buffer, size_t sampleCountPerChunk, uint64_t timeStamp, Function<void()>&& consumeFilledBuffer)
+{
+ std::unique_lock<Lock> lock(m_lock, std::try_to_lock);
+ if (!lock.owns_lock() || !m_ringBuffer)
+ return false;
+
+ ASSERT(buffer.mNumberBuffers == m_ringBuffer->channelCount());
+ if (buffer.mNumberBuffers != m_ringBuffer->channelCount())
+ return false;
+
+ uint64_t startFrame = 0;
+ uint64_t endFrame = 0;
+ m_ringBuffer->getCurrentFrameBounds(startFrame, endFrame);
+ if (timeStamp < startFrame)
+ return false;
+
+ startFrame = timeStamp;
+ while (endFrame - startFrame >= sampleCountPerChunk) {
+ if (m_ringBuffer->fetch(&buffer, sampleCountPerChunk, startFrame, CARingBuffer::Copy))
+ return false;
+ consumeFilledBuffer();
+ startFrame += sampleCountPerChunk;
+ }
+ 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) {
+ size_t byteCount = sampleCount * m_outputDescription->bytesPerFrame();
+ AudioSampleBufferList::zeroABL(buffer, byteCount);
+ 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)
</ins></span></pre>
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