Merge "avoid duplicate of mediaswcodec seccomp policy files"
diff --git a/camera/cameraserver/Android.bp b/camera/cameraserver/Android.bp
index dc7f88a..09a333b 100644
--- a/camera/cameraserver/Android.bp
+++ b/camera/cameraserver/Android.bp
@@ -37,7 +37,7 @@
"android.hardware.camera.device@3.2",
"android.hardware.camera.device@3.4",
],
- compile_multilib: "32",
+ compile_multilib: "prefer32",
cflags: [
"-Wall",
"-Wextra",
diff --git a/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp b/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp
index 3ecf6d5..1495703 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp
+++ b/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp
@@ -148,14 +148,17 @@
// Calculate the output buffer size and determine if any subsamples are
// encrypted.
size_t destSize = 0;
+ size_t srcSize = 0;
bool haveEncryptedSubsamples = false;
for (size_t i = 0; i < subSamples.size(); i++) {
const SubSample &subSample = subSamples[i];
- if (__builtin_add_overflow(destSize, subSample.numBytesOfClearData, &destSize)) {
+ if (__builtin_add_overflow(destSize, subSample.numBytesOfClearData, &destSize) ||
+ __builtin_add_overflow(srcSize, subSample.numBytesOfClearData, &srcSize)) {
_hidl_cb(Status_V1_2::ERROR_DRM_FRAME_TOO_LARGE, 0, "subsample clear size overflow");
return Void();
}
- if (__builtin_add_overflow(destSize, subSample.numBytesOfEncryptedData, &destSize)) {
+ if (__builtin_add_overflow(destSize, subSample.numBytesOfEncryptedData, &destSize) ||
+ __builtin_add_overflow(srcSize, subSample.numBytesOfEncryptedData, &srcSize)) {
_hidl_cb(Status_V1_2::ERROR_DRM_FRAME_TOO_LARGE, 0, "subsample encrypted size overflow");
return Void();
}
@@ -164,7 +167,7 @@
}
}
- if (destSize > destBuffer.size) {
+ if (destSize > destBuffer.size || srcSize > source.size) {
_hidl_cb(Status_V1_2::ERROR_DRM_FRAME_TOO_LARGE, 0, "subsample sum too large");
return Void();
}
diff --git a/media/codec2/components/aac/C2SoftAacDec.cpp b/media/codec2/components/aac/C2SoftAacDec.cpp
index 9884733..677f316 100644
--- a/media/codec2/components/aac/C2SoftAacDec.cpp
+++ b/media/codec2/components/aac/C2SoftAacDec.cpp
@@ -791,7 +791,6 @@
// After an error, replace bufferSize with the sum of the
// decodedSizes to resynchronize the in/out lists.
- inInfo.decodedSizes.pop_back();
inInfo.bufferSize = std::accumulate(
inInfo.decodedSizes.begin(), inInfo.decodedSizes.end(), 0);
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 2699dfe..983887b 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -141,6 +141,7 @@
case AAUDIO_STREAM_STATE_PAUSED:
case AAUDIO_STREAM_STATE_STOPPING:
case AAUDIO_STREAM_STATE_STOPPED:
+ case AAUDIO_STREAM_STATE_FLUSHING:
case AAUDIO_STREAM_STATE_FLUSHED:
break; // Proceed with starting.
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index d50a74b..3bfa2b7 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -342,13 +342,17 @@
// Enable callback before starting AudioRecord to avoid shutting
// down because of a race condition.
mCallbackEnabled.store(true);
+ aaudio_stream_state_t originalState = getState();
+ // Set before starting the callback so that we are in the correct state
+ // before updateStateMachine() can be called by the callback.
+ setState(AAUDIO_STREAM_STATE_STARTING);
mFramesWritten.reset32(); // service writes frames
mTimestampPosition.reset32();
status_t err = mAudioRecord->start(); // resets position to zero
if (err != OK) {
+ mCallbackEnabled.store(false);
+ setState(originalState);
return AAudioConvert_androidToAAudioResult(err);
- } else {
- setState(AAUDIO_STREAM_STATE_STARTING);
}
return AAUDIO_OK;
}
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 74292dd..0427220 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -304,11 +304,15 @@
// Enable callback before starting AudioTrack to avoid shutting
// down because of a race condition.
mCallbackEnabled.store(true);
+ aaudio_stream_state_t originalState = getState();
+ // Set before starting the callback so that we are in the correct state
+ // before updateStateMachine() can be called by the callback.
+ setState(AAUDIO_STREAM_STATE_STARTING);
err = mAudioTrack->start();
if (err != OK) {
+ mCallbackEnabled.store(false);
+ setState(originalState);
return AAudioConvert_androidToAAudioResult(err);
- } else {
- setState(AAUDIO_STREAM_STATE_STARTING);
}
return AAUDIO_OK;
}
diff --git a/media/libaaudio/tests/test_various.cpp b/media/libaaudio/tests/test_various.cpp
index 1c26615..a20c799 100644
--- a/media/libaaudio/tests/test_various.cpp
+++ b/media/libaaudio/tests/test_various.cpp
@@ -124,7 +124,7 @@
}
enum FunctionToCall {
- CALL_START, CALL_STOP, CALL_PAUSE, CALL_FLUSH
+ CALL_START, CALL_STOP, CALL_PAUSE, CALL_FLUSH, CALL_RELEASE
};
void checkStateTransition(aaudio_performance_mode_t perfMode,
@@ -177,11 +177,27 @@
} else if (originalState == AAUDIO_STREAM_STATE_PAUSED) {
ASSERT_EQ(AAUDIO_OK, AAudioStream_requestPause(aaudioStream));
inputState = AAUDIO_STREAM_STATE_PAUSING;
+ } else if (originalState == AAUDIO_STREAM_STATE_FLUSHING) {
+ ASSERT_EQ(AAUDIO_OK, AAudioStream_requestPause(aaudioStream));
+ // We can only flush() after pause is complete.
+ ASSERT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(aaudioStream,
+ AAUDIO_STREAM_STATE_PAUSING,
+ &state,
+ 1000 * NANOS_PER_MILLISECOND));
+ ASSERT_EQ(AAUDIO_STREAM_STATE_PAUSED, state);
+ ASSERT_EQ(AAUDIO_OK, AAudioStream_requestFlush(aaudioStream));
+ // That will put the stream into the FLUSHING state.
+ // The FLUSHING state will persist until we process functionToCall.
+ // That is because the transition to FLUSHED is caused by the callback,
+ // or by calling write() or waitForStateChange(). But those will not
+ // occur.
+ } else if (originalState == AAUDIO_STREAM_STATE_CLOSING) {
+ ASSERT_EQ(AAUDIO_OK, AAudioStream_release(aaudioStream));
}
}
}
- // Wait until past transitional state.
+ // Wait until we get past the transitional state if requested.
if (inputState != AAUDIO_STREAM_STATE_UNINITIALIZED) {
ASSERT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(aaudioStream,
inputState,
@@ -208,12 +224,20 @@
EXPECT_EQ(expectedResult, AAudioStream_requestFlush(aaudioStream));
transitionalState = AAUDIO_STREAM_STATE_FLUSHING;
break;
+ case FunctionToCall::CALL_RELEASE:
+ EXPECT_EQ(expectedResult, AAudioStream_release(aaudioStream));
+ // Set to UNINITIALIZED so the waitForStateChange() below will
+ // will return immediately with the current state.
+ transitionalState = AAUDIO_STREAM_STATE_UNINITIALIZED;
+ break;
}
- EXPECT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(aaudioStream,
- transitionalState,
- &state,
- 1000 * NANOS_PER_MILLISECOND));
+ EXPECT_EQ(AAUDIO_OK,
+ AAudioStream_waitForStateChange(aaudioStream,
+ transitionalState,
+ &state,
+ 1000 * NANOS_PER_MILLISECOND));
+
// We should not change state when a function fails.
if (expectedResult != AAUDIO_OK) {
ASSERT_EQ(originalState, expectedState);
@@ -493,6 +517,88 @@
AAUDIO_STREAM_STATE_FLUSHED);
}
+// FLUSHING ================================================================
+TEST(test_various, aaudio_state_lowlat_flushing_start) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+ AAUDIO_STREAM_STATE_FLUSHING,
+ FunctionToCall::CALL_START,
+ AAUDIO_OK,
+ AAUDIO_STREAM_STATE_STARTED);
+}
+
+TEST(test_various, aaudio_state_none_flushing_start) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+ AAUDIO_STREAM_STATE_FLUSHING,
+ FunctionToCall::CALL_START,
+ AAUDIO_OK,
+ AAUDIO_STREAM_STATE_STARTED);
+}
+
+TEST(test_various, aaudio_state_lowlat_flushing_release) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+ AAUDIO_STREAM_STATE_FLUSHING,
+ FunctionToCall::CALL_RELEASE,
+ AAUDIO_OK,
+ AAUDIO_STREAM_STATE_CLOSING);
+}
+
+TEST(test_various, aaudio_state_none_flushing_release) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+ AAUDIO_STREAM_STATE_FLUSHING,
+ FunctionToCall::CALL_RELEASE,
+ AAUDIO_OK,
+ AAUDIO_STREAM_STATE_CLOSING);
+}
+
+TEST(test_various, aaudio_state_lowlat_starting_release) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+ AAUDIO_STREAM_STATE_STARTING,
+ FunctionToCall::CALL_RELEASE,
+ AAUDIO_OK,
+ AAUDIO_STREAM_STATE_CLOSING);
+}
+
+TEST(test_various, aaudio_state_none_starting_release) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+ AAUDIO_STREAM_STATE_STARTING,
+ FunctionToCall::CALL_RELEASE,
+ AAUDIO_OK,
+ AAUDIO_STREAM_STATE_CLOSING);
+}
+
+// CLOSING ================================================================
+TEST(test_various, aaudio_state_lowlat_closing_start) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+ AAUDIO_STREAM_STATE_CLOSING,
+ FunctionToCall::CALL_START,
+ AAUDIO_ERROR_INVALID_STATE,
+ AAUDIO_STREAM_STATE_CLOSING);
+}
+
+TEST(test_various, aaudio_state_none_closing_start) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+ AAUDIO_STREAM_STATE_CLOSING,
+ FunctionToCall::CALL_START,
+ AAUDIO_ERROR_INVALID_STATE,
+ AAUDIO_STREAM_STATE_CLOSING);
+}
+
+TEST(test_various, aaudio_state_lowlat_closing_stop) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+ AAUDIO_STREAM_STATE_CLOSING,
+ FunctionToCall::CALL_STOP,
+ AAUDIO_ERROR_INVALID_STATE,
+ AAUDIO_STREAM_STATE_CLOSING);
+}
+
+TEST(test_various, aaudio_state_none_closing_stop) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+ AAUDIO_STREAM_STATE_CLOSING,
+ FunctionToCall::CALL_STOP,
+ AAUDIO_ERROR_INVALID_STATE,
+ AAUDIO_STREAM_STATE_CLOSING);
+}
+
// ==========================================================================
TEST(test_various, aaudio_set_buffer_size) {
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 16d2232..6d79aba 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -1001,7 +1001,7 @@
break;
}
- // Whitelist of relevant events to trigger log merging.
+ // List of relevant events that trigger log merging.
// Log merging should activate during audio activity of any kind. This are considered the
// most relevant events.
// TODO should select more wisely the items from the list