commit 6770c6faa3467c92eabc5ec9b23d60eb556a0d03
author Andy Hung <hunga@google.com> Tue Apr 07 13:43:36 2015 -0700
committer Andy Hung <hunga@google.com> Wed Apr 08 15:58:56 2015 -0700
tree 45470127b227dbda5a346bb1902f92f8998c25bb
parent 73c02e4277b399c2ec1555d32b6ad5df23bb83dc

Enable 8 bit and float pcm record formats for AudioFlinger

Update sampling rate handling as well.

Bug: 19570772
Change-Id: I872248e64c0578b2e48869a68fee0d51bd0640c3

services/audioflinger/Threads.cpp

diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index c096bdd..1a20fae 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -86,7 +86,13 @@
 #define ALOGVV(a...) do { } while(0)
 #endif
 
+// TODO: Move these macro/inlines to a header file.
 #define max(a, b) ((a) > (b) ? (a) : (b))
+template <typename T>
+static inline T min(const T& a, const T& b)
+{
+    return a < b ? a : b;
+}
 
 namespace android {
 
@@ -5622,6 +5628,13 @@
                     break;
                 }
 
+                // Don't allow framesOut to be larger than what is possible with resampling
+                // from framesIn.
+                // This isn't strictly necessary but helps limit buffer resizing in
+                // RecordBufferConverter.  TODO: remove when no longer needed.
+                framesOut = min(framesOut,
+                        destinationFramesPossible(
+                                framesIn, mSampleRate, activeTrack->mSampleRate));
                 // process frames from the RecordThread buffer provider to the RecordTrack buffer
                 framesOut = activeTrack->mRecordBufferConverter->convert(
                         activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut);