Title: [216530] trunk/Source/WebCore
- Revision
- 216530
- Author
- commit-qu...@webkit.org
- Date
- 2017-05-09 11:42:47 -0700 (Tue, 09 May 2017)
Log Message
RealtimeOutgoingAudioSource is crashing when given data with more than two channels
https://bugs.webkit.org/show_bug.cgi?id=171868
Patch by Youenn Fablet <you...@apple.com> on 2017-05-09
Reviewed by Eric Carlson.
Manual testing.
* platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp:
(WebCore::libwebrtcAudioFormat): Capping the number of channels to 2.
(WebCore::RealtimeOutgoingAudioSource::audioSamplesAvailable):
Modified Paths
Diff
Modified: trunk/Source/WebCore/ChangeLog (216529 => 216530)
--- trunk/Source/WebCore/ChangeLog 2017-05-09 18:34:27 UTC (rev 216529)
+++ trunk/Source/WebCore/ChangeLog 2017-05-09 18:42:47 UTC (rev 216530)
@@ -1,3 +1,16 @@
+2017-05-09 Youenn Fablet <you...@apple.com>
+
+ RealtimeOutgoingAudioSource is crashing when given data with more than two channels
+ https://bugs.webkit.org/show_bug.cgi?id=171868
+
+ Reviewed by Eric Carlson.
+
+ Manual testing.
+
+ * platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp:
+ (WebCore::libwebrtcAudioFormat): Capping the number of channels to 2.
+ (WebCore::RealtimeOutgoingAudioSource::audioSamplesAvailable):
+
2017-05-09 Myles C. Maxfield <mmaxfi...@apple.com>
font-stretch: normal selects expanded fonts instead of condensed fonts
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp (216529 => 216530)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-05-09 18:34:27 UTC (rev 216529)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-05-09 18:42:47 UTC (rev 216530)
@@ -39,8 +39,10 @@
static inline AudioStreamBasicDescription libwebrtcAudioFormat(Float64 sampleRate, size_t channelCount)
{
+ // FIXME: Microphones can have more than two channels. In such case, we should do the mix down based on AudioChannelLayoutTag.
+ size_t libWebRTCChannelCount = channelCount >= 2 ? 2 : channelCount;
AudioStreamBasicDescription streamFormat;
- FillOutASBDForLPCM(streamFormat, sampleRate, channelCount, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::isFloat, LibWebRTCAudioFormat::isBigEndian, LibWebRTCAudioFormat::isNonInterleaved);
+ FillOutASBDForLPCM(streamFormat, sampleRate, libWebRTCChannelCount, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::isFloat, LibWebRTCAudioFormat::isBigEndian, LibWebRTCAudioFormat::isNonInterleaved);
return streamFormat;
}
@@ -78,8 +80,6 @@
void RealtimeOutgoingAudioSource::audioSamplesAvailable(const MediaTime& time, const PlatformAudioData& audioData, const AudioStreamDescription& streamDescription, size_t sampleCount)
{
- ASSERT(streamDescription.numberOfChannels() <= 2);
-
if (m_inputStreamDescription != streamDescription) {
m_inputStreamDescription = toCAAudioStreamDescription(streamDescription);
auto status = m_sampleConverter->setInputFormat(m_inputStreamDescription);
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