commit 5b10a2037a835e790994b9ebec3c2e55052f1f3b
author Andy Hung <hunga@google.com> Thu Mar 13 13:59:29 2014 -0700
committer Andy Hung <hunga@google.com> Thu Mar 13 15:39:43 2014 -0700
tree f886a67e0d6ae5f9d6d192b76fab7e4bfaa2209a
parent 010a1a1a552cdaad362cea8a0333b8906402dbcb

Fix Offload playback in AudioFlinger

This fixes Offload playback regression introduced by adding
flexible format to mSinkBuffer.  Test case is AAC file playback.

Bug: 13450717
Change-Id: I0fa11978295ed4793be90c565e5b8abedf156914
Signed-off-by: Andy Hung <hunga@google.com>

services/audioflinger/Threads.cpp

diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 8aee194..1c46c27 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1786,8 +1786,9 @@
     // Originally this was int16_t[] array, need to remove legacy implications.
     free(mSinkBuffer);
     mSinkBuffer = NULL;
-    const size_t sinkBufferSize = mNormalFrameCount * mChannelCount
-            * audio_bytes_per_sample(mFormat);
+    // For sink buffer size, we use the frame size from the downstream sink to avoid problems
+    // with non PCM formats for compressed music, e.g. AAC, and Offload threads.
+    const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
     (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
 
     // We resize the mMixerBuffer according to the requirements of the sink buffer which