commit db4c031f518ae5806af73756273ff32cd8d0e4f8
author Andy Hung <hunga@google.com> Wed May 06 08:46:52 2015 -0700
committer Andy Hung <hunga@google.com> Tue May 12 09:30:51 2015 -0700
tree 62f9e0541acccc3acacf808d2a3cdad130eb819b
parent 18aa27016a94d0fee243637a80fd0741f89e08f2

Update sampling rate to 192kHz for devices

Change-Id: I0a83206be51d7ae18ccf85b94b2127356307be69

services/audioflinger/Threads.cpp

diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 7feb63b..5ef017f 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5328,11 +5328,11 @@
         }
         initFastCapture =
                 // either capture sample rate is same as (a reasonable) primary output sample rate
-                (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) &&
+                ((isMusicRate(primaryOutputSampleRate) &&
                     (mSampleRate == primaryOutputSampleRate)) ||
                 // or primary output sample rate is unknown, and capture sample rate is reasonable
                 ((primaryOutputSampleRate == 0) &&
-                    ((mSampleRate == 44100 || mSampleRate == 48000)))) &&
+                        isMusicRate(mSampleRate))) &&
                 // and the buffer size is < 12 ms
                 (mFrameCount * 1000) / mSampleRate < 12;
         break;
@@ -6435,6 +6435,9 @@
         return NO_ERROR;
     }
 
+    ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x"
+            "  srcFormat:%#x dstFormat:%#x  srcRate:%u dstRate:%u",
+            srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate);
     const bool valid =
             audio_is_input_channel(srcChannelMask)
             && audio_is_input_channel(dstChannelMask)