1 files changed, 38 insertions, 11 deletions

diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index a2f4348..1025799 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -69,25 +69,31 @@ status_t AudioTrack::getMinFrameCount(
     //          audio_format_t format
     //          audio_channel_mask_t channelMask
     //          audio_output_flags_t flags
-    int afSampleRate;
+    int afSampleRate = 0;
     if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
         return NO_INIT;
     }
-    int afFrameCount;
+    int afFrameCount = 0;
     if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
         return NO_INIT;
     }
-    uint32_t afLatency;
+    uint32_t afLatency = 0;
     if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
         return NO_INIT;
     }
 
+    if(!afSampleRate || !afFrameCount) {
+        ALOGW("samplerate or framecount 0");
+        return NO_INIT;
+    }
+
     // Ensure that buffer depth covers at least audio hardware latency
     uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
     if (minBufCount < 2) minBufCount = 2;
 
     *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
-            afFrameCount * minBufCount * sampleRate / afSampleRate;
+                afFrameCount * minBufCount * sampleRate / afSampleRate;
+
     ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
             *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
     return NO_ERROR;
@@ -455,11 +461,23 @@ uint32_t AudioTrack::frameCount() const
 
 size_t AudioTrack::frameSize() const
 {
-    if (audio_is_linear_pcm(mFormat)) {
-        return channelCount()*audio_bytes_per_sample(mFormat);
+#ifdef QCOM_HARDWARE
+    if ((audio_stream_type_t)mStreamType == AUDIO_STREAM_VOICE_CALL) {
+       if (audio_is_linear_pcm(mFormat)) {
+          return channelCount()*audio_bytes_per_sample(mFormat);
+       } else {
+          return channelCount()*sizeof(int16_t);
+       }
     } else {
-        return sizeof(uint8_t);
+#endif
+        if (audio_is_linear_pcm(mFormat)) {
+            return channelCount()*audio_bytes_per_sample(mFormat);
+        } else {
+            return sizeof(uint8_t);
+        }
+#ifdef QCOM_HARDWARE
     }
+#endif
 }
 
 sp<IMemory>& AudioTrack::sharedBuffer()
@@ -980,20 +998,26 @@ status_t AudioTrack::createTrack_l(
     } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
 
         // FIXME move these calculations and associated checks to server
-        int afSampleRate;
+        int afSampleRate = 0;
         if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
             return NO_INIT;
         }
-        int afFrameCount;
+        int afFrameCount = 0;
         if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
             return NO_INIT;
         }
 
+        if(!afSampleRate && !afFrameCount) {
+            ALOGW("samplerate or framecount zero");
+            return NO_INIT;
+        }
+
         // Ensure that buffer depth covers at least audio hardware latency
         uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
         if (minBufCount < 2) minBufCount = 2;
 
-        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+        uint32_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+
         ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"
                 ", afLatency=%d",
                 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
@@ -1088,7 +1112,10 @@ status_t AudioTrack::createTrack_l(
     mCblk->waitTimeMs = 0;
     mRemainingFrames = mNotificationFramesAct;
     // FIXME don't believe this lie
-    mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
+    if(sampleRate)
+        mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
+    else
+        mLatency = afLatency;
     // If IAudioTrack is re-created, don't let the requested frameCount
     // decrease.  This can confuse clients that cache frameCount().
     if (mCblk->frameCount > mFrameCount) {