Add some logging in AudioSampleDataConverter::updateBufferedAmount

https://bugs.webkit.org/show_bug.cgi?id=240632

Patch by Youenn Fablet <youennf@gmail.com> on 2022-05-19
Reviewed by Eric Carlson.

This can help debugging audio related issues like echo cancellation.
No change of behavior.

* Source/WebCore/platform/audio/cocoa/AudioSampleDataConverter.mm:
(WebCore::AudioSampleDataConverter::updateBufferedAmount):

Canonical link: https://commits.webkit.org/250736@main
git-svn-id: https://svn.webkit.org/repository/webkit/trunk@294477 268f45cc-cd09-0410-ab3c-d52691b4dbfc

Source/WebCore/platform/audio/cocoa/AudioSampleDataConverter.mm

@@ -27,6 +27,7 @@
 #import "AudioSampleDataConverter.h"
 
 #import "AudioSampleBufferList.h"
+#import "Logging.h"
 #import <AudioToolbox/AudioConverter.h>
 #import <pal/cf/AudioToolboxSoftLink.h>
 
@@ -76,15 +77,30 @@
 {
     if (currentBufferedAmount) {
         if (m_selectedConverter == m_regularConverter) {
-            if (currentBufferedAmount <= m_lowBufferSize)
+            if (currentBufferedAmount <= m_lowBufferSize) {
                 m_selectedConverter = m_lowConverter;
-            else if (currentBufferedAmount >= m_highBufferSize && currentBufferedAmount >= 4 * pushedSampleSize)
+                callOnMainThread([] {
+                    RELEASE_LOG(WebRTC, "AudioSampleDataConverter::updateBufferedAmount low buffer");
+                });
+            } else if (currentBufferedAmount >= m_highBufferSize && currentBufferedAmount >= 4 * pushedSampleSize) {
                 m_selectedConverter = m_highConverter;
+                callOnMainThread([] {
+                    RELEASE_LOG(WebRTC, "AudioSampleDataConverter::updateBufferedAmount high buffer");
+                });
+            }
         } else if (m_selectedConverter == m_highConverter) {
-            if (currentBufferedAmount < m_regularLowBufferSize)
+            if (currentBufferedAmount < m_regularLowBufferSize) {
                 m_selectedConverter = m_regularConverter;
-        } else if (currentBufferedAmount > m_regularHighBufferSize)
+                callOnMainThread([] {
+                    RELEASE_LOG(WebRTC, "AudioSampleDataConverter::updateBufferedAmount going down to regular buffer");
+                });
+            }
+        } else if (currentBufferedAmount > m_regularHighBufferSize) {
             m_selectedConverter = m_regularConverter;
+            callOnMainThread([] {
+                RELEASE_LOG(WebRTC, "AudioSampleDataConverter::updateBufferedAmount going up to regular buffer");
+            });
+        }
     }
     return !!m_selectedConverter;
 }