Diff
Modified: trunk/LayoutTests/ChangeLog (219089 => 219090)
--- trunk/LayoutTests/ChangeLog 2017-07-03 20:17:40 UTC (rev 219089)
+++ trunk/LayoutTests/ChangeLog 2017-07-03 21:01:00 UTC (rev 219090)
@@ -1,3 +1,14 @@
+2017-07-03 Youenn Fablet <[email protected]>
+
+ WebAudioSourceProviderAVFObjC should not reconfigure for each data call
+ https://bugs.webkit.org/show_bug.cgi?id=174101
+
+ Reviewed by Eric Carlson.
+
+ * TestExpectations:
+ * webrtc/peer-connection-audio-mute2.html:
+ * webrtc/peer-connection-remote-audio-mute2.html:
+
2017-07-03 Alex Christensen <[email protected]>
Rebase test after r219024
Modified: trunk/LayoutTests/TestExpectations (219089 => 219090)
--- trunk/LayoutTests/TestExpectations 2017-07-03 20:17:40 UTC (rev 219089)
+++ trunk/LayoutTests/TestExpectations 2017-07-03 21:01:00 UTC (rev 219090)
@@ -774,8 +774,6 @@
webrtc/negotiatedneeded-event-addStream.html [ Pass Crash ]
webrtc/video-replace-track.html [ Pass Failure ]
webrtc/video-getParameters.html [ Failure ]
-webrtc/peer-connection-audio-mute2.html [ Pass Failure ]
-webrtc/peer-connection-remote-audio-mute2.html [ Pass Failure ]
webkit.org/b/170178 webrtc/video-replace-track-to-null.html [ Pass Failure ]
fast/mediastream/RTCPeerConnection-closed-state.html [ Skip ]
fast/mediastream/RTCPeerConnection-iceconnectionstatechange-event.html [ Skip ]
Modified: trunk/LayoutTests/webrtc/peer-connection-audio-mute2.html (219089 => 219090)
--- trunk/LayoutTests/webrtc/peer-connection-audio-mute2.html 2017-07-03 20:17:40 UTC (rev 219089)
+++ trunk/LayoutTests/webrtc/peer-connection-audio-mute2.html 2017-07-03 21:01:00 UTC (rev 219090)
@@ -2,7 +2,7 @@
<html>
<head>
<meta charset="utf-8">
- <title>Testing local audio capture playback causes "playing" event to fire</title>
+ <title>Testing web audio on peer connection audio tracks</title>
<script src=""
<script src=""
</head>
@@ -11,9 +11,6 @@
<script>
var context = new webkitAudioContext();
promise_test((test) => {
- if (window.testRunner)
- testRunner.setUserMediaPermission(true);
-
var localTrack;
return navigator.mediaDevices.getUserMedia({audio: true}).then((localStream) => {
localTrack = localStream.getAudioTracks()[0];
@@ -28,26 +25,20 @@
};
});
}).then(() => {
- return waitFor(500);
+ return doHumAnalysis(remoteStream, true);
+ }).then((value) => {
+ assert_true(value, "heard hum from remote enabled track");
}).then(() => {
- return analyseAudio(remoteStream, 500, context).then((results) => {
- assert_true(results.heardHum, "heard hum from remote enabled track");
- });
- }).then(() => {
localTrack.enabled = false;
- return waitFor(500);
+ return doHumAnalysis(remoteStream, false);
+ }).then((value) => {
+ assert_true(value, "not heard hum from remote disabled track");
}).then(() => {
- return analyseAudio(remoteStream, 500, context).then((results) => {
- assert_false(results.heardHum, "not heard hum from remote disabled track");
- });
- }).then(() => {
localTrack.enabled = true;
- return waitFor(500);
+ return doHumAnalysis(remoteStream, true);
+ }).then((value) => {
+ assert_true(value, "heard hum from remote reenabled track");
}).then(() => {
- return analyseAudio(remoteStream, 500, context).then((results) => {
- assert_true(results.heardHum, "heard hum from remote reenabled track");
- });
- }).then(() => {
return context.close();
});
});
Modified: trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute2.html (219089 => 219090)
--- trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute2.html 2017-07-03 20:17:40 UTC (rev 219089)
+++ trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute2.html 2017-07-03 21:01:00 UTC (rev 219090)
@@ -2,7 +2,7 @@
<html>
<head>
<meta charset="utf-8">
- <title>Testing local audio capture playback causes "playing" event to fire</title>
+ <title>Testing web audio on peer connection audio tracks</title>
<script src=""
<script src=""
</head>
@@ -11,9 +11,6 @@
<script>
var context = new webkitAudioContext();
promise_test((test) => {
- if (window.testRunner)
- testRunner.setUserMediaPermission(true);
-
return navigator.mediaDevices.getUserMedia({audio: true}).then((localStream) => {
var remoteTrack;
var remoteStream;
@@ -28,24 +25,20 @@
};
});
}).then(() => {
- return analyseAudio(remoteStream, 500, context).then((results) => {
- assert_true(results.heardHum, "heard hum from remote enabled track");
- });
+ return doHumAnalysis(remoteStream, true);
+ }).then((value) => {
+ assert_true(value, "heard hum from remote enabled track");
}).then(() => {
remoteTrack.enabled = false;
- return waitFor(100);
+ return doHumAnalysis(remoteStream, false);
+ }).then((value) => {
+ assert_true(value, "not heard hum from remote disabled track");
}).then(() => {
- return analyseAudio(remoteStream, 500, context).then((results) => {
- assert_false(results.heardHum, "not heard hum from remote disabled track");
- });
- }).then(() => {
remoteTrack.enabled = true;
- return waitFor(100);
+ return doHumAnalysis(remoteStream, true);
+ }).then((value) => {
+ assert_true(value, "heard hum from remote reenabled track");
}).then(() => {
- return analyseAudio(remoteStream, 500, context).then((results) => {
- assert_true(results.heardHum, "heard hum from remote reenabled track");
- });
- }).then(() => {
return context.close();
});
});
Modified: trunk/Source/WebCore/ChangeLog (219089 => 219090)
--- trunk/Source/WebCore/ChangeLog 2017-07-03 20:17:40 UTC (rev 219089)
+++ trunk/Source/WebCore/ChangeLog 2017-07-03 21:01:00 UTC (rev 219090)
@@ -1,3 +1,26 @@
+2017-07-03 Youenn Fablet <[email protected]>
+
+ WebAudioSourceProviderAVFObjC should not reconfigure for each data call
+ https://bugs.webkit.org/show_bug.cgi?id=174101
+
+ Reviewed by Eric Carlson.
+
+ Covered by manual testing, in particular
+ https://webrtc.github.io/samples/src/content/peerconnection/webaudio-output/
+ and https://webrtc.github.io/samples/src/content/getusermedia/volume/.
+ Also improved LayoutTests web audio peer connection tests to make them more robust.
+
+ Before the patch, reconfiguration of the web audio provider was happening for every audioSamplesAvailable call.
+ It is now happening only when the format of the audio samples is changing.
+ Changed some member fields from uinque_ptr to optional as a minor improvement.
+
+ * platform/mediastream/mac/WebAudioSourceProviderAVFObjC.h:
+ * platform/mediastream/mac/WebAudioSourceProviderAVFObjC.mm:
+ (WebCore::WebAudioSourceProviderAVFObjC::provideInput):
+ (WebCore::WebAudioSourceProviderAVFObjC::prepare):
+ (WebCore::WebAudioSourceProviderAVFObjC::unprepare):
+ (WebCore::WebAudioSourceProviderAVFObjC::audioSamplesAvailable):
+
2017-06-30 Alex Christensen <[email protected]>
Stop using dispatch_async in ResourceHandleCFURLConnectionDelegateWithOperationQueue
Modified: trunk/Source/WebCore/platform/mediastream/mac/WebAudioSourceProviderAVFObjC.h (219089 => 219090)
--- trunk/Source/WebCore/platform/mediastream/mac/WebAudioSourceProviderAVFObjC.h 2017-07-03 20:17:40 UTC (rev 219089)
+++ trunk/Source/WebCore/platform/mediastream/mac/WebAudioSourceProviderAVFObjC.h 2017-07-03 21:01:00 UTC (rev 219090)
@@ -27,6 +27,7 @@
#if ENABLE(WEB_AUDIO) && ENABLE(MEDIA_STREAM)
+#include "CAAudioStreamDescription.h"
#include "MediaStreamTrackPrivate.h"
#include "WebAudioSourceProvider.h"
#include <CoreAudio/CoreAudioTypes.h>
@@ -50,7 +51,7 @@
static Ref<WebAudioSourceProviderAVFObjC> create(MediaStreamTrackPrivate&);
virtual ~WebAudioSourceProviderAVFObjC();
- void prepare(const AudioStreamBasicDescription*);
+ void prepare(const AudioStreamBasicDescription&);
void unprepare();
private:
@@ -68,8 +69,8 @@
void trackEnabledChanged(MediaStreamTrackPrivate&) final { }
size_t m_listBufferSize { 0 };
- std::unique_ptr<CAAudioStreamDescription> m_inputDescription;
- std::unique_ptr<CAAudioStreamDescription> m_outputDescription;
+ std::optional<CAAudioStreamDescription> m_inputDescription;
+ std::optional<CAAudioStreamDescription> m_outputDescription;
RefPtr<AudioSampleDataSource> m_dataSource;
uint64_t m_writeCount { 0 };
@@ -78,7 +79,6 @@
MediaStreamTrackPrivate* m_captureSource { nullptr };
Lock m_mutex;
bool m_connected { false };
- AudioStreamBasicDescription m_streamFormat;
};
}
Modified: trunk/Source/WebCore/platform/mediastream/mac/WebAudioSourceProviderAVFObjC.mm (219089 => 219090)
--- trunk/Source/WebCore/platform/mediastream/mac/WebAudioSourceProviderAVFObjC.mm 2017-07-03 20:17:40 UTC (rev 219089)
+++ trunk/Source/WebCore/platform/mediastream/mac/WebAudioSourceProviderAVFObjC.mm 2017-07-03 21:01:00 UTC (rev 219090)
@@ -79,7 +79,7 @@
return;
}
- WebAudioBufferList list { *m_outputDescription };
+ WebAudioBufferList list { m_outputDescription.value() };
for (unsigned i = 0; i < bus->numberOfChannels(); ++i) {
AudioChannel& channel = *bus->channel(i);
if (i >= list.bufferCount()) {
@@ -115,15 +115,15 @@
}
}
-void WebAudioSourceProviderAVFObjC::prepare(const AudioStreamBasicDescription* format)
+void WebAudioSourceProviderAVFObjC::prepare(const AudioStreamBasicDescription& format)
{
std::lock_guard<Lock> lock(m_mutex);
LOG(Media, "WebAudioSourceProviderAVFObjC::prepare(%p)", this);
- m_inputDescription = std::make_unique<CAAudioStreamDescription>(*format);
- int numberOfChannels = format->mChannelsPerFrame;
- double sampleRate = format->mSampleRate;
+ m_inputDescription = CAAudioStreamDescription(format);
+ int numberOfChannels = format.mChannelsPerFrame;
+ double sampleRate = format.mSampleRate;
ASSERT(sampleRate >= 0);
const int bytesPerFloat = sizeof(Float32);
@@ -133,12 +133,12 @@
const bool isNonInterleaved = true;
AudioStreamBasicDescription outputDescription { };
FillOutASBDForLPCM(outputDescription, sampleRate, numberOfChannels, bitsPerByte * bytesPerFloat, bitsPerByte * bytesPerFloat, isFloat, isBigEndian, isNonInterleaved);
- m_outputDescription = std::make_unique<CAAudioStreamDescription>(outputDescription);
+ m_outputDescription = CAAudioStreamDescription(outputDescription);
if (!m_dataSource)
m_dataSource = AudioSampleDataSource::create(kRingBufferDuration * sampleRate);
- m_dataSource->setInputFormat(*m_inputDescription);
- m_dataSource->setOutputFormat(*m_outputDescription);
+ m_dataSource->setInputFormat(m_inputDescription.value());
+ m_dataSource->setOutputFormat(m_outputDescription.value());
callOnMainThread([protectedThis = makeRef(*this), numberOfChannels, sampleRate] {
if (protectedThis->m_client)
@@ -150,8 +150,8 @@
{
std::lock_guard<Lock> lock(m_mutex);
- m_inputDescription = nullptr;
- m_outputDescription = nullptr;
+ m_inputDescription = std::nullopt;
+ m_outputDescription = std::nullopt;
m_dataSource = nullptr;
m_listBufferSize = 0;
if (m_captureSource) {
@@ -162,11 +162,10 @@
void WebAudioSourceProviderAVFObjC::audioSamplesAvailable(MediaStreamTrackPrivate&, const MediaTime&, const PlatformAudioData& data, const AudioStreamDescription& description, size_t frameCount)
{
- // FIXME: We should try to call prepare based on trackSettingsChanged callback.
ASSERT(description.platformDescription().type == PlatformDescription::CAAudioStreamBasicType);
auto& basicDescription = *WTF::get<const AudioStreamBasicDescription*>(description.platformDescription().description);
- if (m_streamFormat != basicDescription)
- prepare(&basicDescription);
+ if (!m_inputDescription || m_inputDescription->streamDescription() != basicDescription)
+ prepare(basicDescription);
if (!m_dataSource)
return;
