Diff
Modified: trunk/LayoutTests/ChangeLog (219523 => 219524)
--- trunk/LayoutTests/ChangeLog 2017-07-14 21:26:29 UTC (rev 219523)
+++ trunk/LayoutTests/ChangeLog 2017-07-14 21:43:35 UTC (rev 219524)
@@ -1,3 +1,16 @@
+2017-07-14 Youenn Fablet <you...@apple.com>
+
+ WebRTC: silence data not sent for disabled audio track
+ https://bugs.webkit.org/show_bug.cgi?id=174456
+ <rdar://problem/33284623>
+
+ Reviewed by Eric Carlson.
+
+ * webrtc/audio-muted-stats-expected.txt: Added.
+ * webrtc/audio-muted-stats.html: Added.
+ * webrtc/audio-muted-stats2-expected.txt: Added.
+ * webrtc/audio-muted-stats2.html: Added.
+
2017-07-14 Jonathan Bedard <jbed...@apple.com>
Add High Sierra test expectations
Added: trunk/LayoutTests/webrtc/audio-muted-stats-expected.txt (0 => 219524)
--- trunk/LayoutTests/webrtc/audio-muted-stats-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/audio-muted-stats-expected.txt 2017-07-14 21:43:35 UTC (rev 219524)
@@ -0,0 +1,3 @@
+
+PASS Audio silent data being sent in case track is muted
+
Added: trunk/LayoutTests/webrtc/audio-muted-stats.html (0 => 219524)
--- trunk/LayoutTests/webrtc/audio-muted-stats.html (rev 0)
+++ trunk/LayoutTests/webrtc/audio-muted-stats.html 2017-07-14 21:43:35 UTC (rev 219524)
@@ -0,0 +1,72 @@
+<!doctype html>
+<html>
+ <head>
+ <meta charset="utf-8">
+ <title>Testing basic video exchange from offerer to receiver</title>
+ <script src=""
+ <script src=""
+ </head>
+ <body>
+ <script src =""
+ <script>
+function getOutboundRTPStats(connection)
+{
+ return connection.getStats().then((report) => {
+ var stats;
+ report.forEach((statItem) => {
+ if (statItem.type === "outbound-rtp") {
+ stats = statItem;
+ }
+ });
+ return stats;
+ });
+}
+
+function checkOutboundBytesSentIncreased(firstConnection, statsFirstConnection, count)
+{
+ return getOutboundRTPStats(firstConnection).then((stats) => {
+ if (stats.bytesSent > statsFirstConnection.bytesSent)
+ return;
+ if (++count === 50)
+ return Promise.reject("checking outbound stats bytes sent increasing timed out");
+ return waitFor(50).then(() => {
+ return checkOutboundBytesSentIncreased(firstConnection, statsFirstConnection, count)
+ });
+ });
+}
+
+var track, firstConnection;
+promise_test((test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
+
+ return navigator.mediaDevices.getUserMedia({ audio: true}).then((stream) => {
+ track = stream.getAudioTracks()[0];
+ return new Promise((resolve, reject) => {
+ createConnections((connection) => {
+ firstConnection = connection;
+ firstConnection.addTrack(track, stream);
+ }, (connection) => {
+ connection._ontrack_ = resolve;
+ });
+ setTimeout(() => reject("Test timed out"), 5000);
+ });
+ }).then(() => {
+ return getOutboundRTPStats(firstConnection);
+ }).then((stats) => {
+ statsFirstConnection = stats;
+ return checkOutboundBytesSentIncreased(firstConnection, statsFirstConnection, 0);
+ }).then(() => {
+ track.enabled = false;
+ // Let's wait a little bit so that audio is disabled.
+ return waitFor(100);
+ }).then((stats) => {
+ return getOutboundRTPStats(firstConnection);
+ }).then((stats) => {
+ statsFirstConnection = stats;
+ return checkOutboundBytesSentIncreased(firstConnection, statsFirstConnection, 0);
+ });
+}, "Audio silent data being sent in case track is muted");
+ </script>
+ </body>
+</html>
Added: trunk/LayoutTests/webrtc/audio-muted-stats2-expected.txt (0 => 219524)
--- trunk/LayoutTests/webrtc/audio-muted-stats2-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/audio-muted-stats2-expected.txt 2017-07-14 21:43:35 UTC (rev 219524)
@@ -0,0 +1,3 @@
+
+PASS Audio silent data not being sent in case track is muted from the start
+
Added: trunk/LayoutTests/webrtc/audio-muted-stats2.html (0 => 219524)
--- trunk/LayoutTests/webrtc/audio-muted-stats2.html (rev 0)
+++ trunk/LayoutTests/webrtc/audio-muted-stats2.html 2017-07-14 21:43:35 UTC (rev 219524)
@@ -0,0 +1,64 @@
+<!doctype html>
+<html>
+ <head>
+ <meta charset="utf-8">
+ <title>Testing basic video exchange from offerer to receiver</title>
+ <script src=""
+ <script src=""
+ </head>
+ <body>
+ <script src =""
+ <script>
+function getOutboundRTPStats(connection)
+{
+ return connection.getStats().then((report) => {
+ var stats;
+ report.forEach((statItem) => {
+ if (statItem.type === "outbound-rtp") {
+ stats = statItem;
+ }
+ });
+ return stats;
+ });
+}
+
+function checkOutboundBytesSentNotIncreasing(firstConnection, statsFirstConnection, count)
+{
+ return getOutboundRTPStats(firstConnection).then((stats) => {
+ if (stats.bytesSent > statsFirstConnection.bytesSent)
+ return Promise.reject("outbound stats bytes sent increasing");
+ if (++count === 10)
+ return;
+ return waitFor(50).then(() => {
+ return checkOutboundBytesSentNotIncreasing(firstConnection, statsFirstConnection, count);
+ });
+ });
+}
+
+var track, firstConnection;
+promise_test((test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
+
+ return navigator.mediaDevices.getUserMedia({ audio: true}).then((stream) => {
+ track = stream.getAudioTracks()[0];
+ track.enabled = false;
+ return new Promise((resolve, reject) => {
+ createConnections((connection) => {
+ firstConnection = connection;
+ firstConnection.addTrack(track, stream);
+ }, (connection) => {
+ connection._ontrack_ = resolve;
+ });
+ setTimeout(() => reject("Test timed out"), 5000);
+ });
+ }).then(() => {
+ return getOutboundRTPStats(firstConnection);
+ }).then((stats) => {
+ statsFirstConnection = stats;
+ return checkOutboundBytesSentNotIncreasing(firstConnection, statsFirstConnection, 0);
+ });
+}, "Audio silent data not being sent in case track is muted from the start");
+ </script>
+ </body>
+</html>
Modified: trunk/Source/WebCore/ChangeLog (219523 => 219524)
--- trunk/Source/WebCore/ChangeLog 2017-07-14 21:26:29 UTC (rev 219523)
+++ trunk/Source/WebCore/ChangeLog 2017-07-14 21:43:35 UTC (rev 219524)
@@ -1,3 +1,28 @@
+2017-07-14 Youenn Fablet <you...@apple.com>
+
+ WebRTC: silence data not sent for disabled audio track
+ https://bugs.webkit.org/show_bug.cgi?id=174456
+ <rdar://problem/33284623>
+
+ Reviewed by Eric Carlson.
+
+ Test: webrtc/audio-muted-stats.html
+ webrtc/audio-muted-stats2.html
+
+ Adding a timer-based approach to send 10ms of silence every second.
+ This is consistent with how muted video tracks are implemented.
+ In case the audio track is muted at the time it is added, no silence data is sent.
+
+ * platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp:
+ (WebCore::RealtimeOutgoingAudioSource::RealtimeOutgoingAudioSource):
+ (WebCore::RealtimeOutgoingAudioSource::initializeConverter):
+ (WebCore::RealtimeOutgoingAudioSource::stop):
+ (WebCore::RealtimeOutgoingAudioSource::sourceMutedChanged):
+ (WebCore::RealtimeOutgoingAudioSource::sourceEnabledChanged):
+ (WebCore::RealtimeOutgoingAudioSource::handleMutedIfNeeded):
+ (WebCore::RealtimeOutgoingAudioSource::sendSilence):
+ * platform/mediastream/mac/RealtimeOutgoingAudioSource.h:
+
2017-07-14 Michael Catanzaro <mcatanz...@igalia.com>
[CMake] Unclear distinction between WebKitHelpers and WebKitMacros
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp (219523 => 219524)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-07-14 21:26:29 UTC (rev 219523)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-07-14 21:43:35 UTC (rev 219524)
@@ -49,6 +49,7 @@
RealtimeOutgoingAudioSource::RealtimeOutgoingAudioSource(Ref<MediaStreamTrackPrivate>&& audioSource)
: m_audioSource(WTFMove(audioSource))
, m_sampleConverter(AudioSampleDataSource::create(LibWebRTCAudioFormat::sampleRate * 2))
+ , m_silenceAudioTimer(*this, &RealtimeOutgoingAudioSource::sendSilence)
{
m_audioSource->addObserver(*this);
initializeConverter();
@@ -68,11 +69,12 @@
{
m_muted = m_audioSource->muted();
m_enabled = m_audioSource->enabled();
- m_sampleConverter->setMuted(m_muted || !m_enabled);
+ handleMutedIfNeeded();
}
void RealtimeOutgoingAudioSource::stop()
{
+ m_silenceAudioTimer.stop();
m_audioSource->removeObserver(*this);
}
@@ -79,15 +81,41 @@
void RealtimeOutgoingAudioSource::sourceMutedChanged()
{
m_muted = m_audioSource->muted();
- m_sampleConverter->setMuted(m_muted || !m_enabled);
+ handleMutedIfNeeded();
}
void RealtimeOutgoingAudioSource::sourceEnabledChanged()
{
m_enabled = m_audioSource->enabled();
- m_sampleConverter->setMuted(m_muted || !m_enabled);
+ handleMutedIfNeeded();
}
+void RealtimeOutgoingAudioSource::handleMutedIfNeeded()
+{
+ bool isSilenced = m_muted || !m_enabled;
+ m_sampleConverter->setMuted(isSilenced);
+ if (isSilenced && !m_silenceAudioTimer.isActive())
+ m_silenceAudioTimer.startRepeating(1_s);
+ if (!isSilenced && m_silenceAudioTimer.isActive())
+ m_silenceAudioTimer.stop();
+}
+
+void RealtimeOutgoingAudioSource::sendSilence()
+{
+ LibWebRTCProvider::callOnWebRTCSignalingThread([this, protectedThis = makeRef(*this)] {
+ size_t chunkSampleCount = m_outputStreamDescription.sampleRate() / 100;
+ size_t bufferSize = chunkSampleCount * LibWebRTCAudioFormat::sampleByteSize * m_outputStreamDescription.numberOfChannels();
+
+ if (!bufferSize)
+ return;
+
+ m_audioBuffer.grow(bufferSize);
+ memset(m_audioBuffer.data(), 0, bufferSize);
+ for (auto sink : m_sinks)
+ sink->OnData(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, m_outputStreamDescription.sampleRate(), m_outputStreamDescription.numberOfChannels(), chunkSampleCount);
+ });
+}
+
bool RealtimeOutgoingAudioSource::isReachingBufferedAudioDataHighLimit()
{
auto writtenAudioDuration = m_writeCount / m_inputStreamDescription.sampleRate();
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h (219523 => 219524)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h 2017-07-14 21:26:29 UTC (rev 219523)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h 2017-07-14 21:43:35 UTC (rev 219524)
@@ -33,6 +33,7 @@
#include "AudioSampleDataSource.h"
#include "LibWebRTCMacros.h"
#include "MediaStreamTrackPrivate.h"
+#include "Timer.h"
#include <webrtc/api/mediastreaminterface.h>
#include <wtf/ThreadSafeRefCounted.h>
@@ -83,6 +84,8 @@
void pullAudioData();
void initializeConverter();
+ void handleMutedIfNeeded();
+ void sendSilence();
Vector<webrtc::AudioTrackSinkInterface*> m_sinks;
Ref<MediaStreamTrackPrivate> m_audioSource;
@@ -96,6 +99,7 @@
bool m_muted { false };
bool m_enabled { true };
bool m_skippingAudioData { false };
+ Timer m_silenceAudioTimer;
};
} // namespace WebCore
