Modified: trunk/Source/WebCore/ChangeLog (259631 => 259632)
--- trunk/Source/WebCore/ChangeLog 2020-04-07 09:47:36 UTC (rev 259631)
+++ trunk/Source/WebCore/ChangeLog 2020-04-07 11:14:01 UTC (rev 259632)
@@ -1,3 +1,20 @@
+2020-04-07 Youenn Fablet <[email protected]>
+
+ Remove unnecessary memory allocation from RealtimeIncomingAudioSourceCocoa::OnData
+ https://bugs.webkit.org/show_bug.cgi?id=209969
+
+ Reviewed by Eric Carlson.
+
+ Instead of allocating a new buffer for every audio chunk and copy the audio chunk,
+ Create a WebAudioBufferList once (without any buffer allocation) and set the audio buffer pointer
+ given by libwebrtc as the WebAudioBufferList buffer pointer.
+ We do not take care of muted state anymore since this is done by consumers anyway.
+ Covered by existing tests.
+
+ * platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.cpp:
+ (WebCore::RealtimeIncomingAudioSourceCocoa::OnData):
+ * platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.h:
+
2020-04-07 Claudio Saavedra <[email protected]>
[GTK] gtk_icon_info_free is deprecated since GTK+ 3.8
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.cpp (259631 => 259632)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.cpp 2020-04-07 09:47:36 UTC (rev 259631)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.cpp 2020-04-07 11:14:01 UTC (rev 259632)
@@ -34,8 +34,6 @@
#include "CAAudioStreamDescription.h"
#include "LibWebRTCAudioFormat.h"
#include "Logging.h"
-#include "WebAudioBufferList.h"
-#include "WebAudioSourceProviderAVFObjC.h"
#include <pal/avfoundation/MediaTimeAVFoundation.h>
#include <pal/cf/CoreMediaSoftLink.h>
@@ -69,28 +67,35 @@
void RealtimeIncomingAudioSourceCocoa::OnData(const void* audioData, int bitsPerSample, int sampleRate, size_t numberOfChannels, size_t numberOfFrames)
{
+#if !RELEASE_LOG_DISABLED
+ if (!(++m_chunksReceived % 200)) {
+ callOnMainThread([identifier = LOGIDENTIFIER, this, protectedThis = makeRef(*this), chunksReceived = m_chunksReceived] {
+ ALWAYS_LOG_IF(loggerPtr(), identifier, "chunk ", chunksReceived);
+ });
+ }
+#endif
+
CMTime startTime = CMTimeMake(m_numberOfFrames, sampleRate);
auto mediaTime = PAL::toMediaTime(startTime);
m_numberOfFrames += numberOfFrames;
- AudioStreamBasicDescription newDescription = streamDescription(sampleRate, numberOfChannels);
+ if (!m_audioBufferList || m_sampleRate != sampleRate || m_numberOfChannels != numberOfChannels) {
+ callOnMainThread([identifier = LOGIDENTIFIER, this, protectedThis = makeRef(*this), sampleRate, numberOfChannels] {
+ ALWAYS_LOG_IF(loggerPtr(), identifier, "new audio buffer list for sampleRate ", sampleRate, " and ", numberOfChannels, " channel(s)");
+ });
- // FIXME: We should not need to do the extra memory allocation and copy.
- // Instead, we should be able to directly pass audioData pointer.
- WebAudioBufferList audioBufferList { CAAudioStreamDescription(newDescription), WTF::safeCast<uint32_t>(numberOfFrames) };
- audioBufferList.buffer(0)->mDataByteSize = numberOfChannels * numberOfFrames * bitsPerSample / 8;
- audioBufferList.buffer(0)->mNumberChannels = numberOfChannels;
+ m_sampleRate = sampleRate;
+ m_numberOfChannels = numberOfChannels;
+ m_streamDescription = streamDescription(sampleRate, numberOfChannels);
+ m_audioBufferList = makeUnique<WebAudioBufferList>(m_streamDescription);
+ }
- if (muted())
- memset(audioBufferList.buffer(0)->mData, 0, audioBufferList.buffer(0)->mDataByteSize);
- else
- memcpy(audioBufferList.buffer(0)->mData, audioData, audioBufferList.buffer(0)->mDataByteSize);
+ auto& bufferList = *m_audioBufferList->buffer(0);
+ bufferList.mDataByteSize = numberOfChannels * numberOfFrames * bitsPerSample / 8;
+ bufferList.mNumberChannels = numberOfChannels;
+ bufferList.mData = const_cast<void*>(audioData);
-#if !RELEASE_LOG_DISABLED
- ALWAYS_LOG_IF(loggerPtr() && !(++m_chunksReceived % 200), LOGIDENTIFIER, "chunk ", m_chunksReceived);
-#endif
-
- audioSamplesAvailable(mediaTime, audioBufferList, CAAudioStreamDescription(newDescription), numberOfFrames);
+ audioSamplesAvailable(mediaTime, *m_audioBufferList, m_streamDescription, numberOfFrames);
}
}
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.h (259631 => 259632)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.h 2020-04-07 09:47:36 UTC (rev 259631)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSourceCocoa.h 2020-04-07 11:14:01 UTC (rev 259632)
@@ -30,7 +30,7 @@
#if USE(LIBWEBRTC)
#include "RealtimeIncomingAudioSource.h"
-
+#include "WebAudioBufferList.h"
#include <CoreAudio/CoreAudioTypes.h>
typedef const struct opaqueCMFormatDescription *CMFormatDescriptionRef;
@@ -51,6 +51,10 @@
uint64_t m_numberOfFrames { 0 };
+ int m_sampleRate { 0 };
+ size_t m_numberOfChannels { 0 };
+ CAAudioStreamDescription m_streamDescription;
+ std::unique_ptr<WebAudioBufferList> m_audioBufferList;
#if !RELEASE_LOG_DISABLED
size_t m_chunksReceived { 0 };
#endif
