commit 798ce934571bc7eb5715bf1c03cbe5436a1220e3
author Marco Nelissen <marcone@google.com> Thu May 29 15:35:07 2014 +0000
committer Android (Google) Code Review <android-gerrit@google.com> Thu May 29 15:35:08 2014 +0000
tree aafe457a33a5b221a28a3f00668ac1ed753e12d1
parent 7f692022a6e91723eb84a41f14b64fa565274424
parent eb4860c305def68c5965474bb43c67c2c9d49bbb

Merge "getSampletime -> getSampleTime"

cmds/screenrecord/Overlay.cpp

diff --git a/cmds/screenrecord/Overlay.cpp b/cmds/screenrecord/Overlay.cpp
index 94f560d..c2a8f1b 100644
--- a/cmds/screenrecord/Overlay.cpp
+++ b/cmds/screenrecord/Overlay.cpp
@@ -47,7 +47,7 @@
         "ro.revision",
         "dalvik.vm.heapgrowthlimit",
         "dalvik.vm.heapsize",
-        "persist.sys.dalvik.vm.lib.1",
+        "persist.sys.dalvik.vm.lib.2",
         //"ro.product.cpu.abi",
         //"ro.bootloader",
         //"this-never-appears!",

media/libstagefright/ACodec.cpp

diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 8f154be..d3c508d 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -2872,6 +2872,24 @@
                     break;
                 }
 
+                case OMX_AUDIO_CodingAndroidOPUS:
+                {
+                    OMX_AUDIO_PARAM_ANDROID_OPUSTYPE params;
+                    InitOMXParams(&params);
+                    params.nPortIndex = portIndex;
+
+                    CHECK_EQ((status_t)OK, mOMX->getParameter(
+                            mNode,
+                            (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidOpus,
+                            &params,
+                            sizeof(params)));
+
+                    notify->setString("mime", MEDIA_MIMETYPE_AUDIO_OPUS);
+                    notify->setInt32("channel-count", params.nChannels);
+                    notify->setInt32("sample-rate", params.nSampleRate);
+                    break;
+                }
+
                 default:
                     ALOGE("UNKNOWN AUDIO CODING: %d\n", audioDef->eEncoding);
                     TRESPASS();

services/audioflinger/AudioResamplerDyn.cpp

diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 3abe8fd..a4446a4 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -455,12 +455,13 @@
     const Constants& c(mConstants);
     const TC* const coefs = mConstants.mFirCoefs;
     TI* impulse = mInBuffer.getImpulse();
-    size_t inputIndex = mInputIndex;
+    size_t inputIndex = 0;
     uint32_t phaseFraction = mPhaseFraction;
     const uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;   // stereo output
-    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount) + (phaseFraction != 0);
+    ALOG_ASSERT(0 < inFrameCount && inFrameCount < (1U << 31));
     const uint32_t phaseWrapLimit = c.mL << c.mShift;
 
     // NOTE: be very careful when modifying the code here. register
@@ -474,11 +475,13 @@
         // buffer is empty, fetch a new one
         while (mBuffer.frameCount == 0) {
             mBuffer.frameCount = inFrameCount;
+            ALOG_ASSERT(inFrameCount > 0);
             provider->getNextBuffer(&mBuffer,
                     calculateOutputPTS(outputIndex / 2));
             if (mBuffer.raw == NULL) {
                 goto resample_exit;
             }
+            inFrameCount -= mBuffer.frameCount;
             if (phaseFraction >= phaseWrapLimit) { // read in data
                 mInBuffer.template readAdvance<CHANNELS>(
                         impulse, c.mHalfNumCoefs,
@@ -487,7 +490,7 @@
                 while (phaseFraction >= phaseWrapLimit) {
                     inputIndex++;
                     if (inputIndex >= mBuffer.frameCount) {
-                        inputIndex -= mBuffer.frameCount;
+                        inputIndex = 0;
                         provider->releaseBuffer(&mBuffer);
                         break;
                     }
@@ -535,15 +538,22 @@
 done:
         // often arrives here when input buffer runs out
         if (inputIndex >= frameCount) {
-            inputIndex -= frameCount;
+            inputIndex = 0;
             provider->releaseBuffer(&mBuffer);
-            // mBuffer.frameCount MUST be zero here.
+            ALOG_ASSERT(mBuffer.frameCount == 0);
         }
     }
 
 resample_exit:
+    // Release frames to avoid the count being inaccurate for pts timing.
+    // TODO: Avoid this extra check by making fetch count exact. This is tricky
+    // due to the overfetching mechanism which loads unnecessarily when
+    // mBuffer.frameCount == 0.
+    if (inputIndex) {
+        mBuffer.frameCount = inputIndex;
+        provider->releaseBuffer(&mBuffer);
+    }
     mInBuffer.setImpulse(impulse);
-    mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
 }
 

services/audiopolicy/AudioPolicyManager.cpp

diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
index b047e1d..db0f57d 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -100,6 +100,7 @@
     STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
     STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
     STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
 };
 
 const StringToEnum sFlagNameToEnumTable[] = {
@@ -3164,6 +3165,12 @@
 
     case AUDIO_SOURCE_DEFAULT:
     case AUDIO_SOURCE_MIC:
+    if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
+        device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
+        break;
+    }
+    // FALL THROUGH
+
     case AUDIO_SOURCE_VOICE_RECOGNITION:
     case AUDIO_SOURCE_HOTWORD:
     case AUDIO_SOURCE_VOICE_COMMUNICATION: