Title: [217058] trunk/Source/WebCore
- Revision
- 217058
- Author
- [email protected]
- Date
- 2017-05-18 12:07:55 -0700 (Thu, 18 May 2017)
Log Message
RealtimeOutgoingAudioSource should use the source sample rate
https://bugs.webkit.org/show_bug.cgi?id=172297
Patch by Youenn Fablet <[email protected]> on 2017-05-18
Reviewed by Eric Carlson.
Covered by manual tests.
* platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp:
(WebCore::RealtimeOutgoingAudioSource::audioSamplesAvailable): Using the audio source sample rate so that the converter does the right conversion.
Modified Paths
Diff
Modified: trunk/Source/WebCore/ChangeLog (217057 => 217058)
--- trunk/Source/WebCore/ChangeLog 2017-05-18 18:57:10 UTC (rev 217057)
+++ trunk/Source/WebCore/ChangeLog 2017-05-18 19:07:55 UTC (rev 217058)
@@ -1,3 +1,15 @@
+2017-05-18 Youenn Fablet <[email protected]>
+
+ RealtimeOutgoingAudioSource should use the source sample rate
+ https://bugs.webkit.org/show_bug.cgi?id=172297
+
+ Reviewed by Eric Carlson.
+
+ Covered by manual tests.
+
+ * platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp:
+ (WebCore::RealtimeOutgoingAudioSource::audioSamplesAvailable): Using the audio source sample rate so that the converter does the right conversion.
+
2017-05-18 Andy Estes <[email protected]>
Add "countryCode" to ApplePayErrorContactField
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp (217057 => 217058)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-05-18 18:57:10 UTC (rev 217057)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.cpp 2017-05-18 19:07:55 UTC (rev 217058)
@@ -93,8 +93,10 @@
status = m_sampleConverter->setOutputFormat(m_outputStreamDescription.streamDescription());
ASSERT(!status);
}
-
- m_sampleConverter->pushSamples(MediaTime(m_writeCount, LibWebRTCAudioFormat::sampleRate), audioData, sampleCount);
+
+ // If we change the audio track or its sample rate changes, the timestamp based on m_writeCount may be wrong.
+ // FIXME: We should update m_writeCount to be valid according the new sampleRate.
+ m_sampleConverter->pushSamples(MediaTime(m_writeCount, static_cast<uint32_t>(m_inputStreamDescription.sampleRate())), audioData, sampleCount);
m_writeCount += sampleCount;
LibWebRTCProvider::callOnWebRTCSignalingThread([protectedThis = makeRef(*this)] {
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