Merge "Fix naming convention in CameraService" into sc-dev
diff --git a/drm/mediadrm/plugins/clearkey/default/DrmPlugin.cpp b/drm/mediadrm/plugins/clearkey/default/DrmPlugin.cpp
index 1b8b8c1..6ac3510 100644
--- a/drm/mediadrm/plugins/clearkey/default/DrmPlugin.cpp
+++ b/drm/mediadrm/plugins/clearkey/default/DrmPlugin.cpp
@@ -109,6 +109,7 @@
}
void DrmPlugin::setPlayPolicy() {
+ android::Mutex::Autolock lock(mPlayPolicyLock);
mPlayPolicy.clear();
mPlayPolicy.add(kQueryKeyLicenseType, kStreaming);
mPlayPolicy.add(kQueryKeyPlayAllowed, kTrue);
diff --git a/drm/mediadrm/plugins/clearkey/default/include/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/default/include/DrmPlugin.h
index 4fa42e5..aa9b59d 100644
--- a/drm/mediadrm/plugins/clearkey/default/include/DrmPlugin.h
+++ b/drm/mediadrm/plugins/clearkey/default/include/DrmPlugin.h
@@ -262,6 +262,7 @@
void initProperties();
void setPlayPolicy();
+ android::Mutex mPlayPolicyLock;
android::KeyedVector<String8, String8> mPlayPolicy;
android::KeyedVector<String8, String8> mStringProperties;
android::KeyedVector<String8, Vector<uint8_t>> mByteArrayProperties;
diff --git a/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp b/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp
index f87f830..a77759e 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp
+++ b/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp
@@ -304,6 +304,7 @@
}
void DrmPlugin::setPlayPolicy() {
+ android::Mutex::Autolock lock(mPlayPolicyLock);
mPlayPolicy.clear();
KeyValue policy;
diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
index 3de7589..076beb8 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
@@ -406,6 +406,7 @@
int64_t mCloseSessionOkCount;
int64_t mCloseSessionNotOpenedCount;
uint32_t mNextSecureStopId;
+ android::Mutex mPlayPolicyLock;
// set by property to mock error scenarios
Status_V1_2 mMockError;
diff --git a/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp b/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
index 82c061a..b1cf388 100644
--- a/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
+++ b/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
@@ -30,6 +30,7 @@
namespace android {
constexpr size_t kMinInputBufferSize = 2 * 1024 * 1024;
+constexpr size_t kMaxDimension = 1920;
constexpr char COMPONENT_NAME[] = "c2.android.mpeg2.decoder";
class C2SoftMpeg2Dec::IntfImpl : public SimpleInterface<void>::BaseParams {
@@ -64,8 +65,8 @@
DefineParam(mSize, C2_PARAMKEY_PICTURE_SIZE)
.withDefault(new C2StreamPictureSizeInfo::output(0u, 320, 240))
.withFields({
- C2F(mSize, width).inRange(16, 1920, 4),
- C2F(mSize, height).inRange(16, 1088, 4),
+ C2F(mSize, width).inRange(16, kMaxDimension, 2),
+ C2F(mSize, height).inRange(16, kMaxDimension, 2),
})
.withSetter(SizeSetter)
.build());
@@ -91,8 +92,8 @@
DefineParam(mMaxSize, C2_PARAMKEY_MAX_PICTURE_SIZE)
.withDefault(new C2StreamMaxPictureSizeTuning::output(0u, 320, 240))
.withFields({
- C2F(mSize, width).inRange(2, 1920, 2),
- C2F(mSize, height).inRange(2, 1088, 2),
+ C2F(mSize, width).inRange(2, kMaxDimension, 2),
+ C2F(mSize, height).inRange(2, kMaxDimension, 2),
})
.withSetter(MaxPictureSizeSetter, mSize)
.build());
@@ -204,8 +205,8 @@
const C2P<C2StreamPictureSizeInfo::output> &size) {
(void)mayBlock;
// TODO: get max width/height from the size's field helpers vs. hardcoding
- me.set().width = c2_min(c2_max(me.v.width, size.v.width), 1920u);
- me.set().height = c2_min(c2_max(me.v.height, size.v.height), 1088u);
+ me.set().width = c2_min(c2_max(me.v.width, size.v.width), kMaxDimension);
+ me.set().height = c2_min(c2_max(me.v.height, size.v.height), kMaxDimension);
return C2R::Ok();
}
diff --git a/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp b/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp
index a7cc037..ddd312f 100644
--- a/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp
+++ b/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp
@@ -35,8 +35,10 @@
namespace android {
constexpr size_t kMinInputBufferSize = 2 * 1024 * 1024;
#ifdef MPEG4
+constexpr size_t kMaxDimension = 1920;
constexpr char COMPONENT_NAME[] = "c2.android.mpeg4.decoder";
#else
+constexpr size_t kMaxDimension = 352;
constexpr char COMPONENT_NAME[] = "c2.android.h263.decoder";
#endif
@@ -75,13 +77,8 @@
DefineParam(mSize, C2_PARAMKEY_PICTURE_SIZE)
.withDefault(new C2StreamPictureSizeInfo::output(0u, 176, 144))
.withFields({
-#ifdef MPEG4
- C2F(mSize, width).inRange(2, 1920, 2),
- C2F(mSize, height).inRange(2, 1088, 2),
-#else
- C2F(mSize, width).inRange(2, 352, 2),
- C2F(mSize, height).inRange(2, 288, 2),
-#endif
+ C2F(mSize, width).inRange(2, kMaxDimension, 2),
+ C2F(mSize, height).inRange(2, kMaxDimension, 2),
})
.withSetter(SizeSetter)
.build());
@@ -130,19 +127,10 @@
addParameter(
DefineParam(mMaxSize, C2_PARAMKEY_MAX_PICTURE_SIZE)
-#ifdef MPEG4
- .withDefault(new C2StreamMaxPictureSizeTuning::output(0u, 1920, 1088))
-#else
.withDefault(new C2StreamMaxPictureSizeTuning::output(0u, 352, 288))
-#endif
.withFields({
-#ifdef MPEG4
- C2F(mSize, width).inRange(2, 1920, 2),
- C2F(mSize, height).inRange(2, 1088, 2),
-#else
- C2F(mSize, width).inRange(2, 352, 2),
- C2F(mSize, height).inRange(2, 288, 2),
-#endif
+ C2F(mSize, width).inRange(2, kMaxDimension, 2),
+ C2F(mSize, height).inRange(2, kMaxDimension, 2),
})
.withSetter(MaxPictureSizeSetter, mSize)
.build());
@@ -200,13 +188,8 @@
const C2P<C2StreamPictureSizeInfo::output> &size) {
(void)mayBlock;
// TODO: get max width/height from the size's field helpers vs. hardcoding
-#ifdef MPEG4
- me.set().width = c2_min(c2_max(me.v.width, size.v.width), 1920u);
- me.set().height = c2_min(c2_max(me.v.height, size.v.height), 1088u);
-#else
- me.set().width = c2_min(c2_max(me.v.width, size.v.width), 352u);
- me.set().height = c2_min(c2_max(me.v.height, size.v.height), 288u);
-#endif
+ me.set().width = c2_min(c2_max(me.v.width, size.v.width), kMaxDimension);
+ me.set().height = c2_min(c2_max(me.v.height, size.v.height), kMaxDimension);
return C2R::Ok();
}
diff --git a/media/codec2/sfplugin/Android.bp b/media/codec2/sfplugin/Android.bp
index 94034b5..c3cfcce 100644
--- a/media/codec2/sfplugin/Android.bp
+++ b/media/codec2/sfplugin/Android.bp
@@ -11,6 +11,7 @@
"CCodecConfig.cpp",
"Codec2Buffer.cpp",
"Codec2InfoBuilder.cpp",
+ "FrameReassembler.cpp",
"PipelineWatcher.cpp",
"ReflectedParamUpdater.cpp",
],
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 813d85b..74480b8 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -217,6 +217,7 @@
flags |= C2FrameData::FLAG_CODEC_CONFIG;
}
ALOGV("[%s] queueInputBuffer: buffer->size() = %zu", mName, buffer->size());
+ std::list<std::unique_ptr<C2Work>> items;
std::unique_ptr<C2Work> work(new C2Work);
work->input.ordinal.timestamp = timeUs;
work->input.ordinal.frameIndex = mFrameIndex++;
@@ -226,9 +227,8 @@
work->input.ordinal.customOrdinal = timeUs;
work->input.buffers.clear();
- uint64_t queuedFrameIndex = work->input.ordinal.frameIndex.peeku();
- std::vector<std::shared_ptr<C2Buffer>> queuedBuffers;
sp<Codec2Buffer> copy;
+ bool usesFrameReassembler = false;
if (buffer->size() > 0u) {
Mutexed<Input>::Locked input(mInput);
@@ -253,43 +253,48 @@
"buffer starvation on component.", mName);
}
}
- int32_t cvo = 0;
- if (buffer->meta()->findInt32("cvo", &cvo)) {
- int32_t rotation = cvo % 360;
- // change rotation to counter-clock wise.
- rotation = ((rotation <= 0) ? 0 : 360) - rotation;
- Mutexed<OutputSurface>::Locked output(mOutputSurface);
- output->rotation[queuedFrameIndex] = rotation;
+ if (input->frameReassembler) {
+ usesFrameReassembler = true;
+ input->frameReassembler.process(buffer, &items);
+ } else {
+ int32_t cvo = 0;
+ if (buffer->meta()->findInt32("cvo", &cvo)) {
+ int32_t rotation = cvo % 360;
+ // change rotation to counter-clock wise.
+ rotation = ((rotation <= 0) ? 0 : 360) - rotation;
+
+ Mutexed<OutputSurface>::Locked output(mOutputSurface);
+ uint64_t frameIndex = work->input.ordinal.frameIndex.peeku();
+ output->rotation[frameIndex] = rotation;
+ }
+ work->input.buffers.push_back(c2buffer);
+ if (encryptedBlock) {
+ work->input.infoBuffers.emplace_back(C2InfoBuffer::CreateLinearBuffer(
+ kParamIndexEncryptedBuffer,
+ encryptedBlock->share(0, blockSize, C2Fence())));
+ }
}
- work->input.buffers.push_back(c2buffer);
- if (encryptedBlock) {
- work->input.infoBuffers.emplace_back(C2InfoBuffer::CreateLinearBuffer(
- kParamIndexEncryptedBuffer,
- encryptedBlock->share(0, blockSize, C2Fence())));
- }
- queuedBuffers.push_back(c2buffer);
} else if (eos) {
flags |= C2FrameData::FLAG_END_OF_STREAM;
}
- work->input.flags = (C2FrameData::flags_t)flags;
- // TODO: fill info's
+ if (usesFrameReassembler) {
+ if (!items.empty()) {
+ items.front()->input.configUpdate = std::move(mParamsToBeSet);
+ mFrameIndex = (items.back()->input.ordinal.frameIndex + 1).peek();
+ }
+ } else {
+ work->input.flags = (C2FrameData::flags_t)flags;
+ // TODO: fill info's
- work->input.configUpdate = std::move(mParamsToBeSet);
- work->worklets.clear();
- work->worklets.emplace_back(new C2Worklet);
+ work->input.configUpdate = std::move(mParamsToBeSet);
+ work->worklets.clear();
+ work->worklets.emplace_back(new C2Worklet);
- std::list<std::unique_ptr<C2Work>> items;
- items.push_back(std::move(work));
- mPipelineWatcher.lock()->onWorkQueued(
- queuedFrameIndex,
- std::move(queuedBuffers),
- PipelineWatcher::Clock::now());
- c2_status_t err = mComponent->queue(&items);
- if (err != C2_OK) {
- mPipelineWatcher.lock()->onWorkDone(queuedFrameIndex);
+ items.push_back(std::move(work));
+
+ eos = eos && buffer->size() > 0u;
}
-
- if (err == C2_OK && eos && buffer->size() > 0u) {
+ if (eos) {
work.reset(new C2Work);
work->input.ordinal.timestamp = timeUs;
work->input.ordinal.frameIndex = mFrameIndex++;
@@ -298,23 +303,28 @@
work->input.buffers.clear();
work->input.flags = C2FrameData::FLAG_END_OF_STREAM;
work->worklets.emplace_back(new C2Worklet);
-
- queuedFrameIndex = work->input.ordinal.frameIndex.peeku();
- queuedBuffers.clear();
-
- items.clear();
items.push_back(std::move(work));
-
- mPipelineWatcher.lock()->onWorkQueued(
- queuedFrameIndex,
- std::move(queuedBuffers),
- PipelineWatcher::Clock::now());
- err = mComponent->queue(&items);
- if (err != C2_OK) {
- mPipelineWatcher.lock()->onWorkDone(queuedFrameIndex);
- }
}
- if (err == C2_OK) {
+ c2_status_t err = C2_OK;
+ if (!items.empty()) {
+ {
+ Mutexed<PipelineWatcher>::Locked watcher(mPipelineWatcher);
+ PipelineWatcher::Clock::time_point now = PipelineWatcher::Clock::now();
+ for (const std::unique_ptr<C2Work> &work : items) {
+ watcher->onWorkQueued(
+ work->input.ordinal.frameIndex.peeku(),
+ std::vector(work->input.buffers),
+ now);
+ }
+ }
+ err = mComponent->queue(&items);
+ }
+ if (err != C2_OK) {
+ Mutexed<PipelineWatcher>::Locked watcher(mPipelineWatcher);
+ for (const std::unique_ptr<C2Work> &work : items) {
+ watcher->onWorkDone(work->input.ordinal.frameIndex.peeku());
+ }
+ } else {
Mutexed<Input>::Locked input(mInput);
bool released = false;
if (buffer) {
@@ -946,6 +956,7 @@
bool buffersBoundToCodec) {
C2StreamBufferTypeSetting::input iStreamFormat(0u);
C2StreamBufferTypeSetting::output oStreamFormat(0u);
+ C2ComponentKindSetting kind;
C2PortReorderBufferDepthTuning::output reorderDepth;
C2PortReorderKeySetting::output reorderKey;
C2PortActualDelayTuning::input inputDelay(0);
@@ -957,6 +968,7 @@
{
&iStreamFormat,
&oStreamFormat,
+ &kind,
&reorderDepth,
&reorderKey,
&inputDelay,
@@ -968,7 +980,7 @@
C2_DONT_BLOCK,
nullptr);
if (err == C2_BAD_INDEX) {
- if (!iStreamFormat || !oStreamFormat) {
+ if (!iStreamFormat || !oStreamFormat || !kind) {
return UNKNOWN_ERROR;
}
} else if (err != C2_OK) {
@@ -994,12 +1006,17 @@
if (inputFormat != nullptr) {
bool graphic = (iStreamFormat.value == C2BufferData::GRAPHIC);
+ bool audioEncoder = !graphic && (kind.value == C2Component::KIND_ENCODER);
C2Config::api_feature_t apiFeatures = C2Config::api_feature_t(
API_REFLECTION |
API_VALUES |
API_CURRENT_VALUES |
API_DEPENDENCY |
API_SAME_INPUT_BUFFER);
+ C2StreamAudioFrameSizeInfo::input encoderFrameSize(0u);
+ C2StreamSampleRateInfo::input sampleRate(0u);
+ C2StreamChannelCountInfo::input channelCount(0u);
+ C2StreamPcmEncodingInfo::input pcmEncoding(0u);
std::shared_ptr<C2BlockPool> pool;
{
Mutexed<BlockPools>::Locked pools(mBlockPools);
@@ -1012,7 +1029,19 @@
// from component, create the input block pool with given ID. Otherwise, use default IDs.
std::vector<std::unique_ptr<C2Param>> params;
C2ApiFeaturesSetting featuresSetting{apiFeatures};
- err = mComponent->query({ &featuresSetting },
+ std::vector<C2Param *> stackParams({&featuresSetting});
+ if (audioEncoder) {
+ stackParams.push_back(&encoderFrameSize);
+ stackParams.push_back(&sampleRate);
+ stackParams.push_back(&channelCount);
+ stackParams.push_back(&pcmEncoding);
+ } else {
+ encoderFrameSize.invalidate();
+ sampleRate.invalidate();
+ channelCount.invalidate();
+ pcmEncoding.invalidate();
+ }
+ err = mComponent->query(stackParams,
{ C2PortAllocatorsTuning::input::PARAM_TYPE },
C2_DONT_BLOCK,
¶ms);
@@ -1070,10 +1099,21 @@
input->numSlots = numInputSlots;
input->extraBuffers.flush();
input->numExtraSlots = 0u;
+ if (audioEncoder && encoderFrameSize && sampleRate && channelCount) {
+ input->frameReassembler.init(
+ pool,
+ {C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE},
+ encoderFrameSize.value,
+ sampleRate.value,
+ channelCount.value,
+ pcmEncoding ? pcmEncoding.value : C2Config::PCM_16);
+ }
bool conforming = (apiFeatures & API_SAME_INPUT_BUFFER);
// For encrypted content, framework decrypts source buffer (ashmem) into
// C2Buffers. Thus non-conforming codecs can process these.
- if (!buffersBoundToCodec && (hasCryptoOrDescrambler() || conforming)) {
+ if (!buffersBoundToCodec
+ && !input->frameReassembler
+ && (hasCryptoOrDescrambler() || conforming)) {
input->buffers.reset(new SlotInputBuffers(mName));
} else if (graphic) {
if (mInputSurface) {
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.h b/media/codec2/sfplugin/CCodecBufferChannel.h
index 4dc418a..45da003 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.h
+++ b/media/codec2/sfplugin/CCodecBufferChannel.h
@@ -31,6 +31,7 @@
#include <media/stagefright/CodecBase.h>
#include "CCodecBuffers.h"
+#include "FrameReassembler.h"
#include "InputSurfaceWrapper.h"
#include "PipelineWatcher.h"
@@ -271,6 +272,8 @@
size_t numExtraSlots;
uint32_t inputDelay;
uint32_t pipelineDelay;
+
+ FrameReassembler frameReassembler;
};
Mutexed<Input> mInput;
struct Output {
diff --git a/media/codec2/sfplugin/FrameReassembler.cpp b/media/codec2/sfplugin/FrameReassembler.cpp
new file mode 100644
index 0000000..9cec23f
--- /dev/null
+++ b/media/codec2/sfplugin/FrameReassembler.cpp
@@ -0,0 +1,226 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "FrameReassembler"
+
+#include <log/log.h>
+
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "FrameReassembler.h"
+
+namespace android {
+
+static constexpr uint64_t kToleranceUs = 1000; // 1ms
+
+FrameReassembler::FrameReassembler()
+ : mUsage{0, 0},
+ mSampleRate(0u),
+ mChannelCount(0u),
+ mEncoding(C2Config::PCM_16),
+ mCurrentOrdinal({0, 0, 0}) {
+}
+
+void FrameReassembler::init(
+ const std::shared_ptr<C2BlockPool> &pool,
+ C2MemoryUsage usage,
+ uint32_t frameSize,
+ uint32_t sampleRate,
+ uint32_t channelCount,
+ C2Config::pcm_encoding_t encoding) {
+ mBlockPool = pool;
+ mUsage = usage;
+ mFrameSize = frameSize;
+ mSampleRate = sampleRate;
+ mChannelCount = channelCount;
+ mEncoding = encoding;
+}
+
+void FrameReassembler::updateFrameSize(uint32_t frameSize) {
+ finishCurrentBlock(&mPendingWork);
+ mFrameSize = frameSize;
+}
+
+void FrameReassembler::updateSampleRate(uint32_t sampleRate) {
+ finishCurrentBlock(&mPendingWork);
+ mSampleRate = sampleRate;
+}
+
+void FrameReassembler::updateChannelCount(uint32_t channelCount) {
+ finishCurrentBlock(&mPendingWork);
+ mChannelCount = channelCount;
+}
+
+void FrameReassembler::updatePcmEncoding(C2Config::pcm_encoding_t encoding) {
+ finishCurrentBlock(&mPendingWork);
+ mEncoding = encoding;
+}
+
+void FrameReassembler::reset() {
+ flush();
+ mCurrentOrdinal = {0, 0, 0};
+ mBlockPool.reset();
+ mFrameSize.reset();
+ mSampleRate = 0u;
+ mChannelCount = 0u;
+ mEncoding = C2Config::PCM_16;
+}
+
+FrameReassembler::operator bool() const {
+ return mFrameSize.has_value();
+}
+
+c2_status_t FrameReassembler::process(
+ const sp<MediaCodecBuffer> &buffer,
+ std::list<std::unique_ptr<C2Work>> *items) {
+ int64_t timeUs;
+ if (buffer->size() == 0u
+ || !buffer->meta()->findInt64("timeUs", &timeUs)) {
+ return C2_BAD_VALUE;
+ }
+
+ items->splice(items->end(), mPendingWork);
+
+ // Fill mCurrentBlock
+ if (mCurrentBlock) {
+ // First check the timestamp
+ c2_cntr64_t endTimestampUs = mCurrentOrdinal.timestamp;
+ endTimestampUs += bytesToSamples(mWriteView->size()) * 1000000 / mSampleRate;
+ if (timeUs < endTimestampUs.peek()) {
+ uint64_t diffUs = (endTimestampUs - timeUs).peeku();
+ if (diffUs > kToleranceUs) {
+ // The timestamp is going back in time in large amount.
+ // TODO: b/145702136
+ ALOGW("timestamp going back in time! from %lld to %lld",
+ endTimestampUs.peekll(), (long long)timeUs);
+ }
+ } else { // timeUs >= endTimestampUs.peek()
+ uint64_t diffUs = (timeUs - endTimestampUs).peeku();
+ if (diffUs > kToleranceUs) {
+ // The timestamp is going forward; add silence as necessary.
+ size_t gapSamples = usToSamples(diffUs);
+ size_t remainingSamples =
+ (mWriteView->capacity() - mWriteView->size())
+ / mChannelCount / bytesPerSample();
+ if (gapSamples < remainingSamples) {
+ size_t gapBytes = gapSamples * mChannelCount * bytesPerSample();
+ memset(mWriteView->base() + mWriteView->size(), 0u, gapBytes);
+ mWriteView->setSize(mWriteView->size() + gapBytes);
+ } else {
+ finishCurrentBlock(items);
+ }
+ }
+ }
+ }
+
+ if (mCurrentBlock) {
+ // Append the data at the end of the current block
+ size_t copySize = std::min(
+ buffer->size(),
+ size_t(mWriteView->capacity() - mWriteView->size()));
+ memcpy(mWriteView->base() + mWriteView->size(), buffer->data(), copySize);
+ buffer->setRange(buffer->offset() + copySize, buffer->size() - copySize);
+ mWriteView->setSize(mWriteView->size() + copySize);
+ if (mWriteView->size() == mWriteView->capacity()) {
+ finishCurrentBlock(items);
+ }
+ timeUs += bytesToSamples(copySize) * 1000000 / mSampleRate;
+ }
+
+ if (buffer->size() > 0) {
+ mCurrentOrdinal.timestamp = timeUs;
+ }
+
+ size_t frameSizeBytes = mFrameSize.value() * mChannelCount * bytesPerSample();
+ while (buffer->size() > 0) {
+ LOG_ALWAYS_FATAL_IF(
+ mCurrentBlock,
+ "There's remaining data but the pending block is not filled & finished");
+ std::unique_ptr<C2Work> work(new C2Work);
+ c2_status_t err = mBlockPool->fetchLinearBlock(frameSizeBytes, mUsage, &mCurrentBlock);
+ if (err != C2_OK) {
+ return err;
+ }
+ size_t copySize = std::min(buffer->size(), frameSizeBytes);
+ mWriteView = mCurrentBlock->map().get();
+ if (mWriteView->error() != C2_OK) {
+ return mWriteView->error();
+ }
+ ALOGV("buffer={offset=%zu size=%zu} copySize=%zu",
+ buffer->offset(), buffer->size(), copySize);
+ memcpy(mWriteView->base(), buffer->data(), copySize);
+ mWriteView->setOffset(0u);
+ mWriteView->setSize(copySize);
+ buffer->setRange(buffer->offset() + copySize, buffer->size() - copySize);
+ if (copySize == frameSizeBytes) {
+ finishCurrentBlock(items);
+ }
+ }
+
+ int32_t eos = 0;
+ if (buffer->meta()->findInt32("eos", &eos) && eos) {
+ finishCurrentBlock(items);
+ }
+
+ return C2_OK;
+}
+
+void FrameReassembler::flush() {
+ mPendingWork.clear();
+ mWriteView.reset();
+ mCurrentBlock.reset();
+}
+
+uint64_t FrameReassembler::bytesToSamples(size_t numBytes) const {
+ return numBytes / mChannelCount / bytesPerSample();
+}
+
+size_t FrameReassembler::usToSamples(uint64_t us) const {
+ return (us * mChannelCount * mSampleRate / 1000000);
+}
+
+uint32_t FrameReassembler::bytesPerSample() const {
+ return (mEncoding == C2Config::PCM_8) ? 1
+ : (mEncoding == C2Config::PCM_16) ? 2
+ : (mEncoding == C2Config::PCM_FLOAT) ? 4 : 0;
+}
+
+void FrameReassembler::finishCurrentBlock(std::list<std::unique_ptr<C2Work>> *items) {
+ if (!mCurrentBlock) {
+ // No-op
+ return;
+ }
+ if (mWriteView->size() < mWriteView->capacity()) {
+ memset(mWriteView->base() + mWriteView->size(), 0u,
+ mWriteView->capacity() - mWriteView->size());
+ mWriteView->setSize(mWriteView->capacity());
+ }
+ std::unique_ptr<C2Work> work{std::make_unique<C2Work>()};
+ work->input.ordinal = mCurrentOrdinal;
+ work->input.buffers.push_back(C2Buffer::CreateLinearBuffer(
+ mCurrentBlock->share(0, mCurrentBlock->capacity(), C2Fence())));
+ work->worklets.clear();
+ work->worklets.emplace_back(new C2Worklet);
+ items->push_back(std::move(work));
+
+ ++mCurrentOrdinal.frameIndex;
+ mCurrentOrdinal.timestamp += mFrameSize.value() * 1000000 / mSampleRate;
+ mCurrentBlock.reset();
+ mWriteView.reset();
+}
+
+} // namespace android
diff --git a/media/codec2/sfplugin/FrameReassembler.h b/media/codec2/sfplugin/FrameReassembler.h
new file mode 100644
index 0000000..17ac06d
--- /dev/null
+++ b/media/codec2/sfplugin/FrameReassembler.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef FRAME_REASSEMBLER_H_
+#define FRAME_REASSEMBLER_H_
+
+#include <set>
+#include <memory>
+
+#include <media/MediaCodecBuffer.h>
+
+#include <C2Config.h>
+#include <C2Work.h>
+
+namespace android {
+
+class FrameReassembler {
+public:
+ FrameReassembler();
+
+ void init(
+ const std::shared_ptr<C2BlockPool> &pool,
+ C2MemoryUsage usage,
+ uint32_t frameSize,
+ uint32_t sampleRate,
+ uint32_t channelCount,
+ C2Config::pcm_encoding_t encoding);
+ void updateFrameSize(uint32_t frameSize);
+ void updateSampleRate(uint32_t sampleRate);
+ void updateChannelCount(uint32_t channelCount);
+ void updatePcmEncoding(C2Config::pcm_encoding_t encoding);
+ void reset();
+ void flush();
+
+ explicit operator bool() const;
+
+ c2_status_t process(
+ const sp<MediaCodecBuffer> &buffer,
+ std::list<std::unique_ptr<C2Work>> *items);
+
+private:
+ std::shared_ptr<C2BlockPool> mBlockPool;
+ C2MemoryUsage mUsage;
+ std::optional<uint32_t> mFrameSize;
+ uint32_t mSampleRate;
+ uint32_t mChannelCount;
+ C2Config::pcm_encoding_t mEncoding;
+ std::list<std::unique_ptr<C2Work>> mPendingWork;
+ C2WorkOrdinalStruct mCurrentOrdinal;
+ std::shared_ptr<C2LinearBlock> mCurrentBlock;
+ std::optional<C2WriteView> mWriteView;
+
+ uint64_t bytesToSamples(size_t numBytes) const;
+ size_t usToSamples(uint64_t us) const;
+ uint32_t bytesPerSample() const;
+
+ void finishCurrentBlock(std::list<std::unique_ptr<C2Work>> *items);
+};
+
+} // namespace android
+
+#endif // FRAME_REASSEMBLER_H_
diff --git a/media/codec2/sfplugin/tests/Android.bp b/media/codec2/sfplugin/tests/Android.bp
index 5c774a2..d705cfd 100644
--- a/media/codec2/sfplugin/tests/Android.bp
+++ b/media/codec2/sfplugin/tests/Android.bp
@@ -5,6 +5,7 @@
srcs: [
"CCodecBuffers_test.cpp",
"CCodecConfig_test.cpp",
+ "FrameReassembler_test.cpp",
"ReflectedParamUpdater_test.cpp",
],
diff --git a/media/codec2/sfplugin/tests/FrameReassembler_test.cpp b/media/codec2/sfplugin/tests/FrameReassembler_test.cpp
new file mode 100644
index 0000000..6738ee7
--- /dev/null
+++ b/media/codec2/sfplugin/tests/FrameReassembler_test.cpp
@@ -0,0 +1,340 @@
+/*
+ * Copyright 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "FrameReassembler.h"
+
+#include <gtest/gtest.h>
+
+#include <C2PlatformSupport.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+namespace android {
+
+static size_t BytesPerSample(C2Config::pcm_encoding_t encoding) {
+ return encoding == PCM_8 ? 1
+ : encoding == PCM_16 ? 2
+ : encoding == PCM_FLOAT ? 4 : 0;
+}
+
+static uint64_t Diff(c2_cntr64_t a, c2_cntr64_t b) {
+ return std::abs((a - b).peek());
+}
+
+class FrameReassemblerTest : public ::testing::Test {
+public:
+ static const C2MemoryUsage kUsage;
+ static constexpr uint64_t kTimestampToleranceUs = 100;
+
+ FrameReassemblerTest() {
+ mInitStatus = GetCodec2BlockPool(C2BlockPool::BASIC_LINEAR, nullptr, &mPool);
+ }
+
+ status_t initStatus() const { return mInitStatus; }
+
+ void testPushSameSize(
+ size_t encoderFrameSize,
+ size_t sampleRate,
+ size_t channelCount,
+ C2Config::pcm_encoding_t encoding,
+ size_t inputFrameSizeInBytes,
+ size_t count,
+ size_t expectedOutputSize) {
+ FrameReassembler frameReassembler;
+ frameReassembler.init(
+ mPool,
+ kUsage,
+ encoderFrameSize,
+ sampleRate,
+ channelCount,
+ encoding);
+
+ ASSERT_TRUE(frameReassembler) << "FrameReassembler init failed";
+
+ size_t inputIndex = 0, outputIndex = 0;
+ size_t expectCount = 0;
+ for (size_t i = 0; i < count; ++i) {
+ sp<MediaCodecBuffer> buffer = new MediaCodecBuffer(
+ new AMessage, new ABuffer(inputFrameSizeInBytes));
+ buffer->setRange(0, inputFrameSizeInBytes);
+ buffer->meta()->setInt64(
+ "timeUs",
+ inputIndex * 1000000 / sampleRate / channelCount / BytesPerSample(encoding));
+ if (i == count - 1) {
+ buffer->meta()->setInt32("eos", 1);
+ }
+ for (size_t j = 0; j < inputFrameSizeInBytes; ++j, ++inputIndex) {
+ buffer->base()[j] = (inputIndex & 0xFF);
+ }
+ std::list<std::unique_ptr<C2Work>> items;
+ ASSERT_EQ(C2_OK, frameReassembler.process(buffer, &items));
+ while (!items.empty()) {
+ std::unique_ptr<C2Work> work = std::move(*items.begin());
+ items.erase(items.begin());
+ // Verify timestamp
+ uint64_t expectedTimeUs =
+ outputIndex * 1000000 / sampleRate / channelCount / BytesPerSample(encoding);
+ EXPECT_GE(
+ kTimestampToleranceUs,
+ Diff(expectedTimeUs, work->input.ordinal.timestamp))
+ << "expected timestamp: " << expectedTimeUs
+ << " actual timestamp: " << work->input.ordinal.timestamp.peeku()
+ << " output index: " << outputIndex;
+
+ // Verify buffer
+ ASSERT_EQ(1u, work->input.buffers.size());
+ std::shared_ptr<C2Buffer> buffer = work->input.buffers.front();
+ ASSERT_EQ(C2BufferData::LINEAR, buffer->data().type());
+ ASSERT_EQ(1u, buffer->data().linearBlocks().size());
+ C2ReadView view = buffer->data().linearBlocks().front().map().get();
+ ASSERT_EQ(C2_OK, view.error());
+ ASSERT_EQ(encoderFrameSize * BytesPerSample(encoding), view.capacity());
+ for (size_t j = 0; j < view.capacity(); ++j, ++outputIndex) {
+ ASSERT_TRUE(outputIndex < inputIndex
+ || inputIndex == inputFrameSizeInBytes * count);
+ uint8_t expected = outputIndex < inputIndex ? (outputIndex & 0xFF) : 0;
+ if (expectCount < 10) {
+ ++expectCount;
+ EXPECT_EQ(expected, view.data()[j]) << "output index = " << outputIndex;
+ }
+ }
+ }
+ }
+
+ ASSERT_EQ(inputFrameSizeInBytes * count, inputIndex);
+ size_t encoderFrameSizeInBytes =
+ encoderFrameSize * channelCount * BytesPerSample(encoding);
+ ASSERT_EQ(0, outputIndex % encoderFrameSizeInBytes)
+ << "output size must be multiple of frame size: output size = " << outputIndex
+ << " frame size = " << encoderFrameSizeInBytes;
+ ASSERT_EQ(expectedOutputSize, outputIndex)
+ << "output size must be smallest multiple of frame size, "
+ << "equal to or larger than input size. output size = " << outputIndex
+ << " input size = " << inputIndex << " frame size = " << encoderFrameSizeInBytes;
+ }
+
+private:
+ status_t mInitStatus;
+ std::shared_ptr<C2BlockPool> mPool;
+};
+
+const C2MemoryUsage FrameReassemblerTest::kUsage{C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE};
+
+// Push frames with exactly the same size as the encoder requested.
+TEST_F(FrameReassemblerTest, PushExactFrameSize) {
+ ASSERT_EQ(OK, initStatus());
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_8,
+ 1024 /* input frame size in bytes = 1024 samples * 1 channel * 1 bytes/sample */,
+ 10 /* count */,
+ 10240 /* expected output size = 10 * 1024 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_16,
+ 2048 /* input frame size in bytes = 1024 samples * 1 channel * 2 bytes/sample */,
+ 10 /* count */,
+ 20480 /* expected output size = 10 * 2048 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_FLOAT,
+ 4096 /* input frame size in bytes = 1024 samples * 1 channel * 4 bytes/sample */,
+ 10 /* count */,
+ 40960 /* expected output size = 10 * 4096 bytes/frame */);
+}
+
+// Push frames with half the size that the encoder requested.
+TEST_F(FrameReassemblerTest, PushHalfFrameSize) {
+ ASSERT_EQ(OK, initStatus());
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_8,
+ 512 /* input frame size in bytes = 512 samples * 1 channel * 1 bytes per sample */,
+ 10 /* count */,
+ 5120 /* expected output size = 5 * 1024 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_16,
+ 1024 /* input frame size in bytes = 512 samples * 1 channel * 2 bytes per sample */,
+ 10 /* count */,
+ 10240 /* expected output size = 5 * 2048 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_FLOAT,
+ 2048 /* input frame size in bytes = 512 samples * 1 channel * 4 bytes per sample */,
+ 10 /* count */,
+ 20480 /* expected output size = 5 * 4096 bytes/frame */);
+}
+
+// Push frames with twice the size that the encoder requested.
+TEST_F(FrameReassemblerTest, PushDoubleFrameSize) {
+ ASSERT_EQ(OK, initStatus());
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_8,
+ 2048 /* input frame size in bytes = 2048 samples * 1 channel * 1 bytes per sample */,
+ 10 /* count */,
+ 20480 /* expected output size = 20 * 1024 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_16,
+ 4096 /* input frame size in bytes = 2048 samples * 1 channel * 2 bytes per sample */,
+ 10 /* count */,
+ 40960 /* expected output size = 20 * 2048 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_FLOAT,
+ 8192 /* input frame size in bytes = 2048 samples * 1 channel * 4 bytes per sample */,
+ 10 /* count */,
+ 81920 /* expected output size = 20 * 4096 bytes/frame */);
+}
+
+// Push frames with a little bit larger (+5 samples) than the requested size.
+TEST_F(FrameReassemblerTest, PushLittleLargerFrameSize) {
+ ASSERT_EQ(OK, initStatus());
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_8,
+ 1029 /* input frame size in bytes = 1029 samples * 1 channel * 1 bytes per sample */,
+ 10 /* count */,
+ 11264 /* expected output size = 11 * 1024 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_16,
+ 2058 /* input frame size in bytes = 1029 samples * 1 channel * 2 bytes per sample */,
+ 10 /* count */,
+ 22528 /* expected output size = 11 * 2048 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_FLOAT,
+ 4116 /* input frame size in bytes = 1029 samples * 1 channel * 4 bytes per sample */,
+ 10 /* count */,
+ 45056 /* expected output size = 11 * 4096 bytes/frame */);
+}
+
+// Push frames with a little bit smaller (-5 samples) than the requested size.
+TEST_F(FrameReassemblerTest, PushLittleSmallerFrameSize) {
+ ASSERT_EQ(OK, initStatus());
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_8,
+ 1019 /* input frame size in bytes = 1019 samples * 1 channel * 1 bytes per sample */,
+ 10 /* count */,
+ 10240 /* expected output size = 10 * 1024 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_16,
+ 2038 /* input frame size in bytes = 1019 samples * 1 channel * 2 bytes per sample */,
+ 10 /* count */,
+ 20480 /* expected output size = 10 * 2048 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_FLOAT,
+ 4076 /* input frame size in bytes = 1019 samples * 1 channel * 4 bytes per sample */,
+ 10 /* count */,
+ 40960 /* expected output size = 10 * 4096 bytes/frame */);
+}
+
+// Push single-byte frames
+TEST_F(FrameReassemblerTest, PushSingleByte) {
+ ASSERT_EQ(OK, initStatus());
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_8,
+ 1 /* input frame size in bytes */,
+ 100000 /* count */,
+ 100352 /* expected output size = 98 * 1024 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_16,
+ 1 /* input frame size in bytes */,
+ 100000 /* count */,
+ 100352 /* expected output size = 49 * 2048 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_FLOAT,
+ 1 /* input frame size in bytes */,
+ 100000 /* count */,
+ 102400 /* expected output size = 25 * 4096 bytes/frame */);
+}
+
+// Push one big chunk.
+TEST_F(FrameReassemblerTest, PushBigChunk) {
+ ASSERT_EQ(OK, initStatus());
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_8,
+ 100000 /* input frame size in bytes */,
+ 1 /* count */,
+ 100352 /* expected output size = 98 * 1024 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_16,
+ 100000 /* input frame size in bytes */,
+ 1 /* count */,
+ 100352 /* expected output size = 49 * 2048 bytes/frame */);
+ testPushSameSize(
+ 1024 /* frame size in samples */,
+ 48000 /* sample rate */,
+ 1 /* channel count */,
+ PCM_FLOAT,
+ 100000 /* input frame size in bytes */,
+ 1 /* count */,
+ 102400 /* expected output size = 25 * 4096 bytes/frame */);
+}
+
+} // namespace android
diff --git a/media/libaudioclient/fuzzer/Android.bp b/media/libaudioclient/fuzzer/Android.bp
new file mode 100644
index 0000000..edfc5a5
--- /dev/null
+++ b/media/libaudioclient/fuzzer/Android.bp
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_fuzz {
+ name: "audioflinger_fuzzer",
+ srcs: [
+ "audioflinger_fuzzer.cpp",
+ ],
+ static_libs: [
+ "android.hardware.audio.common@7.0-enums",
+ "effect-aidl-unstable-cpp",
+ "libaudioclient",
+ "libbase",
+ "libcgrouprc",
+ "libcgrouprc_format",
+ "libcutils",
+ "libjsoncpp",
+ "liblog",
+ "libmediametrics",
+ "libmediametricsservice",
+ "libmedia_helper",
+ "libprocessgroup",
+ "shared-file-region-aidl-unstable-cpp",
+ ],
+ shared_libs: [
+ "android.hardware.audio.common-util",
+ "audioclient-types-aidl-unstable-cpp",
+ "audioflinger-aidl-unstable-cpp",
+ "audiopolicy-aidl-unstable-cpp",
+ "audiopolicy-types-aidl-unstable-cpp",
+ "av-types-aidl-unstable-cpp",
+ "capture_state_listener-aidl-unstable-cpp",
+ "libaudioclient_aidl_conversion",
+ "libaudioflinger",
+ "libaudiofoundation",
+ "libaudiomanager",
+ "libaudiopolicy",
+ "libaudioutils",
+ "libbinder",
+ "libdl",
+ "libmediautils",
+ "libnblog",
+ "libutils",
+ "libxml2",
+ "mediametricsservice-aidl-unstable-cpp",
+ ],
+ header_libs: [
+ "libaudiofoundation_headers",
+ "libmedia_headers",
+ ],
+ fuzz_config: {
+ cc: [
+ "android-media-fuzzing-reports@google.com",
+ ],
+ componentid: 155276,
+ },
+}
diff --git a/media/libaudioclient/fuzzer/README.md b/media/libaudioclient/fuzzer/README.md
new file mode 100644
index 0000000..ada6c49
--- /dev/null
+++ b/media/libaudioclient/fuzzer/README.md
@@ -0,0 +1,80 @@
+# Fuzzer for libaudioflinger
+
+## Plugin Design Considerations
+The fuzzer plugin for libaudioflinger is designed based on the understanding of the
+library and tries to achieve the following:
+
+##### Maximize code coverage
+The configuration parameters are not hardcoded, but instead selected based on
+incoming data. This ensures more code paths are reached by the fuzzer. The fuzzer
+covers libaudioflinger APIs as called from libaudioclient through IPC.
+
+libaudioflinger supports the following parameters:
+1. Unique IDs (parameter name: `uniqueId`)
+2. Audio Mode (parameter name: `mode`)
+3. Session ID (parameter name: `sessionId`)
+4. Encapsulation Mode (parameter name: `encapsulationMode`)
+5. Audio Port Role (parameter name: `portRole`)
+6. Audio Port Type (parameter name: `portType`)
+7. Audio Stream Type (parameter name: `streamType`)
+8. Audio Format (parameter name: `format`)
+9. Audio Channel Mask (parameter name: `channelMask`)
+10. Usage (parameter name: `usage`)
+11. Audio Content Type (parameter name: `contentType`)
+12. Input Source (parameter name: `inputSource`)
+13. Input Flags (parameter name: `inputFlags`)
+14. Output Flags (parameter name: `outputFlags`)
+15. Audio Gain Mode (parameter name: `gainMode`)
+16. Audio Device (parameter name: `device`)
+
+| Parameter| Valid Values| Configured Value|
+|------------- |-------------| ----- |
+| `uniqueId` | 0. `AUDIO_UNIQUE_ID_USE_UNSPECIFIED` 1. `AUDIO_UNIQUE_ID_USE_SESSION` 2. `AUDIO_UNIQUE_ID_USE_MODULE` 3. `AUDIO_UNIQUE_ID_USE_EFFECT` 4. `AUDIO_UNIQUE_ID_USE_PATCH` 5. `AUDIO_UNIQUE_ID_USE_OUTPUT` 6. `AUDIO_UNIQUE_ID_USE_INPUT` 7. `AUDIO_UNIQUE_ID_USE_CLIENT` 8. `AUDIO_UNIQUE_ID_USE_MAX` | Value obtained from FuzzedDataProvider
+| `mode` | 0.`AUDIO_MODE_INVALID` 1. `AUDIO_MODE_CURRENT` 2. ` AUDIO_MODE_NORMAL` 3. `AUDIO_MODE_RINGTONE` 4. `AUDIO_MODE_IN_CALL` 5. `AUDIO_MODE_IN_COMMUNICATION` 6. `AUDIO_MODE_CALL_SCREEN` | Value obtained from FuzzedDataProvider|
+| `sessionId` | 0. `AUDIO_SESSION_NONE` 1. `AUDIO_SESSION_OUTPUT_STAGE` 2. `AUDIO_SESSION_DEVICE` | Value obtained from FuzzedDataProvider|
+| `encapsulationMode` | 0. `AUDIO_ENCAPSULATION_MODE_NONE` 1. `AUDIO_ENCAPSULATION_MODE_ELEMENTARY_STREAM` 2. `AUDIO_ENCAPSULATION_MODE_HANDLE` | Value obtained from FuzzedDataProvider|
+| `portRole` | 0. `AUDIO_PORT_ROLE_NONE` 1. `AUDIO_PORT_ROLE_SOURCE` 2. `AUDIO_PORT_ROLE_SINK` | Value obtained from FuzzedDataProvider|
+| `portType` | 0. `AUDIO_PORT_TYPE_NONE` 1. `AUDIO_PORT_TYPE_DEVICE` 2. `AUDIO_PORT_TYPE_MIX` 3. `AUDIO_PORT_TYPE_SESSION`| Value obtained from FuzzedDataProvider|
+| `streamType` | 15 values of type `audio_stream_type_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `format` | 77 values of type `audio_format_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `channelMask` | 83 values of type `audio_channel_mask_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `usage` | 22 values of type `audio_usage_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `contentType` | 5 values of type `audio_content_type_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `inputSource` | 14 values of type `audio_source_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `inputFlags` | 9 values of type `audio_input_flags_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `outputFlags` | 16 values of type `audio_output_flags_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `gainMode` | 3 values of type `audio_gain_mode_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `device` | 66 values of type `audio_devices_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+
+This also ensures that the plugin is always deterministic for any given input.
+
+##### Maximize utilization of input data
+The plugin tolerates any kind of input (empty, huge,
+malformed, etc) and doesn't `exit()` on any input and thereby increasing the
+chance of identifying vulnerabilities.
+
+## Build
+
+This describes steps to build audioflinger_fuzzer binary.
+
+### Android
+
+#### Steps to build
+Build the fuzzer
+```
+ $ mm -j$(nproc) audioflinger_fuzzer
+```
+
+#### Steps to run
+Create a directory CORPUS_DIR and copy some files to that folder
+Push this directory to device.
+
+To run on device
+```
+ $ adb sync data
+ $ adb shell /data/fuzz/arm64/audioflinger_fuzzer/audioflinger_fuzzer CORPUS_DIR
+```
+
+## References:
+ * http://llvm.org/docs/LibFuzzer.html
+ * https://github.co
diff --git a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
new file mode 100644
index 0000000..db2b0b8
--- /dev/null
+++ b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
@@ -0,0 +1,730 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ */
+
+/**
+ * NOTE
+ * 1) The input to AudioFlinger binder calls are fuzzed in this fuzzer
+ * 2) AudioFlinger crashes due to the fuzzer are detected by the
+ Binder DeathRecipient, where the fuzzer aborts if AudioFlinger dies
+ */
+
+#include <android_audio_policy_configuration_V7_0-enums.h>
+#include <binder/IServiceManager.h>
+#include <binder/MemoryDealer.h>
+#include <media/AudioEffect.h>
+#include <media/AudioRecord.h>
+#include <media/AudioSystem.h>
+#include <media/AudioTrack.h>
+#include <media/IAudioFlinger.h>
+#include "fuzzer/FuzzedDataProvider.h"
+
+#define MAX_STRING_LENGTH 256
+#define MAX_ARRAY_LENGTH 256
+
+using namespace std;
+using namespace android;
+
+namespace xsd {
+using namespace ::android::audio::policy::configuration::V7_0;
+}
+
+constexpr audio_unique_id_use_t kUniqueIds[] = {
+ AUDIO_UNIQUE_ID_USE_UNSPECIFIED, AUDIO_UNIQUE_ID_USE_SESSION, AUDIO_UNIQUE_ID_USE_MODULE,
+ AUDIO_UNIQUE_ID_USE_EFFECT, AUDIO_UNIQUE_ID_USE_PATCH, AUDIO_UNIQUE_ID_USE_OUTPUT,
+ AUDIO_UNIQUE_ID_USE_INPUT, AUDIO_UNIQUE_ID_USE_CLIENT, AUDIO_UNIQUE_ID_USE_MAX,
+};
+
+constexpr audio_mode_t kModes[] = {
+ AUDIO_MODE_INVALID, AUDIO_MODE_CURRENT, AUDIO_MODE_NORMAL, AUDIO_MODE_RINGTONE,
+ AUDIO_MODE_IN_CALL, AUDIO_MODE_IN_COMMUNICATION, AUDIO_MODE_CALL_SCREEN};
+
+constexpr audio_session_t kSessionId[] = {AUDIO_SESSION_NONE, AUDIO_SESSION_OUTPUT_STAGE,
+ AUDIO_SESSION_DEVICE};
+
+constexpr audio_encapsulation_mode_t kEncapsulation[] = {
+ AUDIO_ENCAPSULATION_MODE_NONE,
+ AUDIO_ENCAPSULATION_MODE_ELEMENTARY_STREAM,
+ AUDIO_ENCAPSULATION_MODE_HANDLE,
+};
+
+constexpr audio_port_role_t kPortRoles[] = {
+ AUDIO_PORT_ROLE_NONE,
+ AUDIO_PORT_ROLE_SOURCE,
+ AUDIO_PORT_ROLE_SINK,
+};
+
+constexpr audio_port_type_t kPortTypes[] = {
+ AUDIO_PORT_TYPE_NONE,
+ AUDIO_PORT_TYPE_DEVICE,
+ AUDIO_PORT_TYPE_MIX,
+ AUDIO_PORT_TYPE_SESSION,
+};
+
+template <typename T, typename X, typename FUNC>
+std::vector<T> getFlags(const xsdc_enum_range<X> &range, const FUNC &func,
+ const std::string &findString = {}) {
+ std::vector<T> vec;
+ for (const auto &xsdEnumVal : range) {
+ T enumVal;
+ std::string enumString = toString(xsdEnumVal);
+ if (enumString.find(findString) != std::string::npos &&
+ func(enumString.c_str(), &enumVal)) {
+ vec.push_back(enumVal);
+ }
+ }
+ return vec;
+}
+
+static const std::vector<audio_stream_type_t> kStreamtypes =
+ getFlags<audio_stream_type_t, xsd::AudioStreamType, decltype(audio_stream_type_from_string)>(
+ xsdc_enum_range<xsd::AudioStreamType>{}, audio_stream_type_from_string);
+
+static const std::vector<audio_format_t> kFormats =
+ getFlags<audio_format_t, xsd::AudioFormat, decltype(audio_format_from_string)>(
+ xsdc_enum_range<xsd::AudioFormat>{}, audio_format_from_string);
+
+static const std::vector<audio_channel_mask_t> kChannelMasks =
+ getFlags<audio_channel_mask_t, xsd::AudioChannelMask, decltype(audio_channel_mask_from_string)>(
+ xsdc_enum_range<xsd::AudioChannelMask>{}, audio_channel_mask_from_string);
+
+static const std::vector<audio_usage_t> kUsages =
+ getFlags<audio_usage_t, xsd::AudioUsage, decltype(audio_usage_from_string)>(
+ xsdc_enum_range<xsd::AudioUsage>{}, audio_usage_from_string);
+
+static const std::vector<audio_content_type_t> kContentType =
+ getFlags<audio_content_type_t, xsd::AudioContentType, decltype(audio_content_type_from_string)>(
+ xsdc_enum_range<xsd::AudioContentType>{}, audio_content_type_from_string);
+
+static const std::vector<audio_source_t> kInputSources =
+ getFlags<audio_source_t, xsd::AudioSource, decltype(audio_source_from_string)>(
+ xsdc_enum_range<xsd::AudioSource>{}, audio_source_from_string);
+
+static const std::vector<audio_gain_mode_t> kGainModes =
+ getFlags<audio_gain_mode_t, xsd::AudioGainMode, decltype(audio_gain_mode_from_string)>(
+ xsdc_enum_range<xsd::AudioGainMode>{}, audio_gain_mode_from_string);
+
+static const std::vector<audio_devices_t> kDevices =
+ getFlags<audio_devices_t, xsd::AudioDevice, decltype(audio_device_from_string)>(
+ xsdc_enum_range<xsd::AudioDevice>{}, audio_device_from_string);
+
+static const std::vector<audio_input_flags_t> kInputFlags =
+ getFlags<audio_input_flags_t, xsd::AudioInOutFlag, decltype(audio_input_flag_from_string)>(
+ xsdc_enum_range<xsd::AudioInOutFlag>{}, audio_input_flag_from_string, "_INPUT_");
+
+static const std::vector<audio_output_flags_t> kOutputFlags =
+ getFlags<audio_output_flags_t, xsd::AudioInOutFlag, decltype(audio_output_flag_from_string)>(
+ xsdc_enum_range<xsd::AudioInOutFlag>{}, audio_output_flag_from_string, "_OUTPUT_");
+
+template <typename T, size_t size>
+T getValueFromArray(FuzzedDataProvider *fdp, const T (&arr)[size]) {
+ return arr[fdp->ConsumeIntegralInRange<int32_t>(0, size - 1)];
+}
+
+template <typename T, size_t size>
+T getValue(FuzzedDataProvider *fdp, const T (&arr)[size]) {
+ if (fdp->ConsumeBool()) {
+ return static_cast<T>(fdp->ConsumeIntegral<int32_t>());
+ }
+ return getValueFromArray(fdp, arr);
+}
+
+template <typename T>
+T getValueFromVector(FuzzedDataProvider *fdp, std::vector<T> vec) {
+ return vec[fdp->ConsumeIntegralInRange<int32_t>(0, vec.size() - 1)];
+}
+
+template <typename T>
+T getValue(FuzzedDataProvider *fdp, std::vector<T> vec) {
+ if (fdp->ConsumeBool()) {
+ return static_cast<T>(fdp->ConsumeIntegral<int32_t>());
+ }
+ return getValueFromVector(fdp, vec);
+}
+
+class DeathNotifier : public IBinder::DeathRecipient {
+ public:
+ void binderDied(const wp<IBinder> &) { abort(); }
+};
+
+class AudioFlingerFuzzer {
+ public:
+ AudioFlingerFuzzer(const uint8_t *data, size_t size);
+ void process();
+
+ private:
+ FuzzedDataProvider mFdp;
+ void invokeAudioTrack();
+ void invokeAudioRecord();
+ status_t invokeAudioEffect();
+ void invokeAudioSystem();
+ status_t invokeAudioInputDevice();
+ status_t invokeAudioOutputDevice();
+ void invokeAudioPatch();
+
+ sp<DeathNotifier> mDeathNotifier;
+};
+
+AudioFlingerFuzzer::AudioFlingerFuzzer(const uint8_t *data, size_t size) : mFdp(data, size) {
+ sp<IServiceManager> sm = defaultServiceManager();
+ sp<IBinder> binder = sm->getService(String16("media.audio_flinger"));
+ if (binder == nullptr) {
+ return;
+ }
+ mDeathNotifier = new DeathNotifier();
+ binder->linkToDeath(mDeathNotifier);
+}
+
+void AudioFlingerFuzzer::invokeAudioTrack() {
+ uint32_t sampleRate = mFdp.ConsumeIntegral<uint32_t>();
+ audio_format_t format = getValue(&mFdp, kFormats);
+ audio_channel_mask_t channelMask = getValue(&mFdp, kChannelMasks);
+ size_t frameCount = static_cast<size_t>(mFdp.ConsumeIntegral<uint32_t>());
+ int32_t notificationFrames = mFdp.ConsumeIntegral<int32_t>();
+ uint32_t useSharedBuffer = mFdp.ConsumeBool();
+ audio_output_flags_t flags = getValue(&mFdp, kOutputFlags);
+ audio_session_t sessionId = getValue(&mFdp, kSessionId);
+ audio_usage_t usage = getValue(&mFdp, kUsages);
+ audio_content_type_t contentType = getValue(&mFdp, kContentType);
+ audio_attributes_t attributes = {};
+ sp<IMemory> sharedBuffer;
+ sp<MemoryDealer> heap = nullptr;
+ audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
+
+ bool offload = false;
+ bool fast = ((flags & AUDIO_OUTPUT_FLAG_FAST) != 0);
+
+ if (useSharedBuffer != 0) {
+ size_t heapSize = audio_channel_count_from_out_mask(channelMask) *
+ audio_bytes_per_sample(format) * frameCount;
+ heap = new MemoryDealer(heapSize, "AudioTrack Heap Base");
+ sharedBuffer = heap->allocate(heapSize);
+ frameCount = 0;
+ notificationFrames = 0;
+ }
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ offloadInfo.sample_rate = sampleRate;
+ offloadInfo.channel_mask = channelMask;
+ offloadInfo.format = format;
+ offload = true;
+ }
+
+ attributes.content_type = contentType;
+ attributes.usage = usage;
+ sp<AudioTrack> track = new AudioTrack();
+
+ track->set(AUDIO_STREAM_DEFAULT, sampleRate, format, channelMask, frameCount, flags, nullptr,
+ nullptr, notificationFrames, sharedBuffer, false, sessionId,
+ ((fast && sharedBuffer == 0) || offload) ? AudioTrack::TRANSFER_CALLBACK
+ : AudioTrack::TRANSFER_DEFAULT,
+ offload ? &offloadInfo : nullptr, getuid(), getpid(), &attributes, false, 1.0f,
+ AUDIO_PORT_HANDLE_NONE);
+
+ status_t status = track->initCheck();
+ if (status != NO_ERROR) {
+ track.clear();
+ return;
+ }
+ track->getSampleRate();
+ track->latency();
+ track->getUnderrunCount();
+ track->streamType();
+ track->channelCount();
+ track->getNotificationPeriodInFrames();
+ uint32_t bufferSizeInFrames = mFdp.ConsumeIntegral<uint32_t>();
+ track->setBufferSizeInFrames(bufferSizeInFrames);
+ track->getBufferSizeInFrames();
+
+ int64_t duration = mFdp.ConsumeIntegral<int64_t>();
+ track->getBufferDurationInUs(&duration);
+ sp<IMemory> sharedBuffer2 = track->sharedBuffer();
+ track->setCallerName(mFdp.ConsumeRandomLengthString(MAX_STRING_LENGTH));
+
+ track->setVolume(mFdp.ConsumeFloatingPoint<float>(), mFdp.ConsumeFloatingPoint<float>());
+ track->setVolume(mFdp.ConsumeFloatingPoint<float>());
+ track->setAuxEffectSendLevel(mFdp.ConsumeFloatingPoint<float>());
+
+ float auxEffectSendLevel;
+ track->getAuxEffectSendLevel(&auxEffectSendLevel);
+ track->setSampleRate(mFdp.ConsumeIntegral<uint32_t>());
+ track->getSampleRate();
+ track->getOriginalSampleRate();
+
+ AudioPlaybackRate playbackRate = {};
+ playbackRate.mSpeed = mFdp.ConsumeFloatingPoint<float>();
+ playbackRate.mPitch = mFdp.ConsumeFloatingPoint<float>();
+ track->setPlaybackRate(playbackRate);
+ track->getPlaybackRate();
+ track->setLoop(mFdp.ConsumeIntegral<uint32_t>(), mFdp.ConsumeIntegral<uint32_t>(),
+ mFdp.ConsumeIntegral<uint32_t>());
+ track->setMarkerPosition(mFdp.ConsumeIntegral<uint32_t>());
+
+ uint32_t marker = {};
+ track->getMarkerPosition(&marker);
+ track->setPositionUpdatePeriod(mFdp.ConsumeIntegral<uint32_t>());
+
+ uint32_t updatePeriod = {};
+ track->getPositionUpdatePeriod(&updatePeriod);
+ track->setPosition(mFdp.ConsumeIntegral<uint32_t>());
+ uint32_t position = {};
+ track->getPosition(&position);
+ track->getBufferPosition(&position);
+ track->reload();
+ track->start();
+ track->pause();
+ track->flush();
+ track->stop();
+ track->stopped();
+}
+
+void AudioFlingerFuzzer::invokeAudioRecord() {
+ int32_t notificationFrames = mFdp.ConsumeIntegral<int32_t>();
+ uint32_t sampleRate = mFdp.ConsumeIntegral<uint32_t>();
+ size_t frameCount = static_cast<size_t>(mFdp.ConsumeIntegral<uint32_t>());
+ audio_format_t format = getValue(&mFdp, kFormats);
+ audio_channel_mask_t channelMask = getValue(&mFdp, kChannelMasks);
+ audio_input_flags_t flags = getValue(&mFdp, kInputFlags);
+ audio_session_t sessionId = getValue(&mFdp, kSessionId);
+ audio_source_t inputSource = getValue(&mFdp, kInputSources);
+
+ audio_attributes_t attributes = {};
+ bool fast = ((flags & AUDIO_OUTPUT_FLAG_FAST) != 0);
+
+ attributes.source = inputSource;
+
+ sp<AudioRecord> record = new AudioRecord(String16(mFdp.ConsumeRandomLengthString().c_str()));
+ record->set(AUDIO_SOURCE_DEFAULT, sampleRate, format, channelMask, frameCount, nullptr, nullptr,
+ notificationFrames, false, sessionId,
+ fast ? AudioRecord::TRANSFER_CALLBACK : AudioRecord::TRANSFER_DEFAULT, flags,
+ getuid(), getpid(), &attributes, AUDIO_PORT_HANDLE_NONE);
+ status_t status = record->initCheck();
+ if (status != NO_ERROR) {
+ return;
+ }
+ record->latency();
+ record->format();
+ record->channelCount();
+ record->frameCount();
+ record->frameSize();
+ record->inputSource();
+ record->getNotificationPeriodInFrames();
+ record->start();
+ record->stop();
+ record->stopped();
+
+ uint32_t marker = mFdp.ConsumeIntegral<uint32_t>();
+ record->setMarkerPosition(marker);
+ record->getMarkerPosition(&marker);
+
+ uint32_t updatePeriod = mFdp.ConsumeIntegral<uint32_t>();
+ record->setPositionUpdatePeriod(updatePeriod);
+ record->getPositionUpdatePeriod(&updatePeriod);
+
+ uint32_t position;
+ record->getPosition(&position);
+
+ ExtendedTimestamp timestamp;
+ record->getTimestamp(×tamp);
+ record->getSessionId();
+ record->getCallerName();
+ android::AudioRecord::Buffer audioBuffer;
+ int32_t waitCount = mFdp.ConsumeIntegral<int32_t>();
+ size_t nonContig = static_cast<size_t>(mFdp.ConsumeIntegral<uint32_t>());
+ audioBuffer.frameCount = static_cast<size_t>(mFdp.ConsumeIntegral<uint32_t>());
+ record->obtainBuffer(&audioBuffer, waitCount, &nonContig);
+ bool blocking = false;
+ record->read(audioBuffer.raw, audioBuffer.size, blocking);
+ record->getInputFramesLost();
+ record->getFlags();
+
+ std::vector<media::MicrophoneInfo> activeMicrophones;
+ record->getActiveMicrophones(&activeMicrophones);
+ record->releaseBuffer(&audioBuffer);
+
+ audio_port_handle_t deviceId =
+ static_cast<audio_port_handle_t>(mFdp.ConsumeIntegral<int32_t>());
+ record->setInputDevice(deviceId);
+ record->getInputDevice();
+ record->getRoutedDeviceId();
+ record->getPortId();
+}
+
+struct EffectClient : public android::media::BnEffectClient {
+ EffectClient() {}
+ binder::Status controlStatusChanged(bool controlGranted __unused) override {
+ return binder::Status::ok();
+ }
+ binder::Status enableStatusChanged(bool enabled __unused) override {
+ return binder::Status::ok();
+ }
+ binder::Status commandExecuted(int32_t cmdCode __unused,
+ const std::vector<uint8_t> &cmdData __unused,
+ const std::vector<uint8_t> &replyData __unused) override {
+ return binder::Status::ok();
+ }
+};
+
+status_t AudioFlingerFuzzer::invokeAudioEffect() {
+ effect_uuid_t type;
+ type.timeLow = mFdp.ConsumeIntegral<uint32_t>();
+ type.timeMid = mFdp.ConsumeIntegral<uint16_t>();
+ type.timeHiAndVersion = mFdp.ConsumeIntegral<uint16_t>();
+ type.clockSeq = mFdp.ConsumeIntegral<uint16_t>();
+ for (int i = 0; i < 6; ++i) {
+ type.node[i] = mFdp.ConsumeIntegral<uint8_t>();
+ }
+
+ effect_descriptor_t descriptor = {};
+ descriptor.type = type;
+ descriptor.uuid = *EFFECT_UUID_NULL;
+
+ sp<EffectClient> effectClient(new EffectClient());
+
+ const int32_t priority = mFdp.ConsumeIntegral<int32_t>();
+ audio_session_t sessionId = static_cast<audio_session_t>(mFdp.ConsumeIntegral<int32_t>());
+ const audio_io_handle_t io = mFdp.ConsumeIntegral<int32_t>();
+ String16 opPackageName = static_cast<String16>(mFdp.ConsumeRandomLengthString().c_str());
+ AudioDeviceTypeAddr device;
+
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (!af) {
+ return NO_ERROR;
+ }
+
+ media::CreateEffectRequest request{};
+ request.desc =
+ VALUE_OR_RETURN_STATUS(legacy2aidl_effect_descriptor_t_EffectDescriptor(descriptor));
+ request.client = effectClient;
+ request.priority = priority;
+ request.output = io;
+ request.sessionId = sessionId;
+ request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(device));
+ request.opPackageName = VALUE_OR_RETURN_STATUS(legacy2aidl_String16_string(opPackageName));
+ request.pid = getpid();
+ request.probe = false;
+
+ media::CreateEffectResponse response{};
+ status_t status = af->createEffect(request, &response);
+
+ if (status != OK) {
+ return NO_ERROR;
+ }
+
+ descriptor =
+ VALUE_OR_RETURN_STATUS(aidl2legacy_EffectDescriptor_effect_descriptor_t(response.desc));
+
+ uint32_t numEffects;
+ af->queryNumberEffects(&numEffects);
+
+ uint32_t queryIndex = mFdp.ConsumeIntegral<uint32_t>();
+ af->queryEffect(queryIndex, &descriptor);
+
+ effect_descriptor_t getDescriptor;
+ uint32_t preferredTypeFlag = mFdp.ConsumeIntegral<int32_t>();
+ af->getEffectDescriptor(&descriptor.uuid, &descriptor.type, preferredTypeFlag, &getDescriptor);
+
+ sessionId = static_cast<audio_session_t>(mFdp.ConsumeIntegral<int32_t>());
+ audio_io_handle_t srcOutput = mFdp.ConsumeIntegral<int32_t>();
+ audio_io_handle_t dstOutput = mFdp.ConsumeIntegral<int32_t>();
+ af->moveEffects(sessionId, srcOutput, dstOutput);
+
+ int effectId = mFdp.ConsumeIntegral<int32_t>();
+ sessionId = static_cast<audio_session_t>(mFdp.ConsumeIntegral<int32_t>());
+ af->setEffectSuspended(effectId, sessionId, mFdp.ConsumeBool());
+ return NO_ERROR;
+}
+
+void AudioFlingerFuzzer::invokeAudioSystem() {
+ AudioSystem::muteMicrophone(mFdp.ConsumeBool());
+ AudioSystem::setMasterMute(mFdp.ConsumeBool());
+ AudioSystem::setMasterVolume(mFdp.ConsumeFloatingPoint<float>());
+ AudioSystem::setMasterBalance(mFdp.ConsumeFloatingPoint<float>());
+ AudioSystem::setVoiceVolume(mFdp.ConsumeFloatingPoint<float>());
+
+ float volume;
+ AudioSystem::getMasterVolume(&volume);
+
+ bool state;
+ AudioSystem::getMasterMute(&state);
+ AudioSystem::isMicrophoneMuted(&state);
+
+ audio_stream_type_t stream = getValue(&mFdp, kStreamtypes);
+ AudioSystem::setStreamMute(getValue(&mFdp, kStreamtypes), mFdp.ConsumeBool());
+
+ stream = getValue(&mFdp, kStreamtypes);
+ AudioSystem::setStreamVolume(stream, mFdp.ConsumeFloatingPoint<float>(),
+ mFdp.ConsumeIntegral<int32_t>());
+
+ audio_mode_t mode = getValue(&mFdp, kModes);
+ AudioSystem::setMode(mode);
+
+ size_t frameCount;
+ stream = getValue(&mFdp, kStreamtypes);
+ AudioSystem::getOutputFrameCount(&frameCount, stream);
+
+ uint32_t latency;
+ stream = getValue(&mFdp, kStreamtypes);
+ AudioSystem::getOutputLatency(&latency, stream);
+
+ stream = getValue(&mFdp, kStreamtypes);
+ AudioSystem::getStreamVolume(stream, &volume, mFdp.ConsumeIntegral<int32_t>());
+
+ stream = getValue(&mFdp, kStreamtypes);
+ AudioSystem::getStreamMute(stream, &state);
+
+ uint32_t samplingRate;
+ AudioSystem::getSamplingRate(mFdp.ConsumeIntegral<int32_t>(), &samplingRate);
+
+ AudioSystem::getFrameCount(mFdp.ConsumeIntegral<int32_t>(), &frameCount);
+ AudioSystem::getLatency(mFdp.ConsumeIntegral<int32_t>(), &latency);
+ AudioSystem::setVoiceVolume(mFdp.ConsumeFloatingPoint<float>());
+
+ uint32_t halFrames;
+ uint32_t dspFrames;
+ AudioSystem::getRenderPosition(mFdp.ConsumeIntegral<int32_t>(), &halFrames, &dspFrames);
+
+ AudioSystem::getInputFramesLost(mFdp.ConsumeIntegral<int32_t>());
+ AudioSystem::getInputFramesLost(mFdp.ConsumeIntegral<int32_t>());
+
+ audio_unique_id_use_t uniqueIdUse = getValue(&mFdp, kUniqueIds);
+ AudioSystem::newAudioUniqueId(uniqueIdUse);
+
+ audio_session_t sessionId = getValue(&mFdp, kSessionId);
+ pid_t pid = mFdp.ConsumeBool() ? getpid() : mFdp.ConsumeIntegral<int32_t>();
+ uid_t uid = mFdp.ConsumeBool() ? getuid() : mFdp.ConsumeIntegral<int32_t>();
+ AudioSystem::acquireAudioSessionId(sessionId, pid, uid);
+
+ pid = mFdp.ConsumeBool() ? getpid() : mFdp.ConsumeIntegral<int32_t>();
+ sessionId = getValue(&mFdp, kSessionId);
+ AudioSystem::releaseAudioSessionId(sessionId, pid);
+
+ sessionId = getValue(&mFdp, kSessionId);
+ AudioSystem::getAudioHwSyncForSession(sessionId);
+
+ AudioSystem::systemReady();
+ AudioSystem::getFrameCountHAL(mFdp.ConsumeIntegral<int32_t>(), &frameCount);
+
+ size_t buffSize;
+ uint32_t sampleRate = mFdp.ConsumeIntegral<uint32_t>();
+ audio_format_t format = getValue(&mFdp, kFormats);
+ audio_channel_mask_t channelMask = getValue(&mFdp, kChannelMasks);
+ AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &buffSize);
+
+ AudioSystem::getPrimaryOutputSamplingRate();
+ AudioSystem::getPrimaryOutputFrameCount();
+ AudioSystem::setLowRamDevice(mFdp.ConsumeBool(), mFdp.ConsumeIntegral<int64_t>());
+
+ std::vector<media::MicrophoneInfo> microphones;
+ AudioSystem::getMicrophones(µphones);
+
+ std::vector<pid_t> pids;
+ pids.insert(pids.begin(), getpid());
+ for (int i = 1; i < mFdp.ConsumeIntegralInRange<int32_t>(2, MAX_ARRAY_LENGTH); ++i) {
+ pids.insert(pids.begin() + i, static_cast<pid_t>(mFdp.ConsumeIntegral<int32_t>()));
+ }
+ AudioSystem::setAudioHalPids(pids);
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (!af) {
+ return;
+ }
+ af->setRecordSilenced(mFdp.ConsumeIntegral<uint32_t>(), mFdp.ConsumeBool());
+
+ float balance = mFdp.ConsumeFloatingPoint<float>();
+ af->getMasterBalance(&balance);
+ af->invalidateStream(static_cast<audio_stream_type_t>(mFdp.ConsumeIntegral<uint32_t>()));
+}
+
+status_t AudioFlingerFuzzer::invokeAudioInputDevice() {
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (!af) {
+ return NO_ERROR;
+ }
+
+ audio_config_t config = {};
+ audio_module_handle_t module = mFdp.ConsumeIntegral<int32_t>();
+ audio_io_handle_t input = mFdp.ConsumeIntegral<int32_t>();
+ config.frame_count = mFdp.ConsumeIntegral<uint32_t>();
+ String8 address = static_cast<String8>(mFdp.ConsumeRandomLengthString().c_str());
+
+ config.channel_mask = getValue(&mFdp, kChannelMasks);
+ config.format = getValue(&mFdp, kFormats);
+
+ config.offload_info = AUDIO_INFO_INITIALIZER;
+ config.offload_info.bit_rate = mFdp.ConsumeIntegral<uint32_t>();
+ config.offload_info.bit_width = mFdp.ConsumeIntegral<uint32_t>();
+ config.offload_info.content_id = mFdp.ConsumeIntegral<uint32_t>();
+ config.offload_info.channel_mask = getValue(&mFdp, kChannelMasks);
+ config.offload_info.duration_us = mFdp.ConsumeIntegral<int64_t>();
+ config.offload_info.encapsulation_mode = getValue(&mFdp, kEncapsulation);
+ config.offload_info.format = getValue(&mFdp, kFormats);
+ config.offload_info.has_video = mFdp.ConsumeBool();
+ config.offload_info.is_streaming = mFdp.ConsumeBool();
+ config.offload_info.sample_rate = (mFdp.ConsumeIntegral<uint32_t>());
+ config.offload_info.sync_id = mFdp.ConsumeIntegral<uint32_t>();
+ config.offload_info.stream_type = getValue(&mFdp, kStreamtypes);
+ config.offload_info.usage = getValue(&mFdp, kUsages);
+
+ config.sample_rate = mFdp.ConsumeIntegral<uint32_t>();
+
+ audio_devices_t device = getValue(&mFdp, kDevices);
+ audio_source_t source = getValue(&mFdp, kInputSources);
+ audio_input_flags_t flags = getValue(&mFdp, kInputFlags);
+
+ AudioDeviceTypeAddr deviceTypeAddr(device, address.c_str());
+
+ media::OpenInputRequest request{};
+ request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
+ request.input = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(input));
+ request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+ request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(deviceTypeAddr));
+ request.source = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_source_t_AudioSourceType(source));
+ request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_input_flags_t_int32_t_mask(flags));
+
+ media::OpenInputResponse response{};
+ status_t status = af->openInput(request, &response);
+ if (status != NO_ERROR) {
+ return NO_ERROR;
+ }
+
+ input = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_module_handle_t(response.input));
+ af->closeInput(input);
+ return NO_ERROR;
+}
+
+status_t AudioFlingerFuzzer::invokeAudioOutputDevice() {
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (!af) {
+ return NO_ERROR;
+ }
+
+ audio_config_t config = {};
+ audio_module_handle_t module = mFdp.ConsumeIntegral<int32_t>();
+ audio_io_handle_t output = mFdp.ConsumeIntegral<int32_t>();
+ config.frame_count = mFdp.ConsumeIntegral<uint32_t>();
+ String8 address = static_cast<String8>(mFdp.ConsumeRandomLengthString().c_str());
+
+ config.channel_mask = getValue(&mFdp, kChannelMasks);
+
+ config.offload_info = AUDIO_INFO_INITIALIZER;
+ config.offload_info.bit_rate = mFdp.ConsumeIntegral<uint32_t>();
+ config.offload_info.bit_width = mFdp.ConsumeIntegral<uint32_t>();
+ config.offload_info.channel_mask = getValue(&mFdp, kChannelMasks);
+ config.offload_info.content_id = mFdp.ConsumeIntegral<uint32_t>();
+ config.offload_info.duration_us = mFdp.ConsumeIntegral<int64_t>();
+ config.offload_info.encapsulation_mode = getValue(&mFdp, kEncapsulation);
+ config.offload_info.format = getValue(&mFdp, kFormats);
+ config.offload_info.has_video = mFdp.ConsumeBool();
+ config.offload_info.is_streaming = mFdp.ConsumeBool();
+ config.offload_info.sample_rate = mFdp.ConsumeIntegral<uint32_t>();
+ config.offload_info.stream_type = getValue(&mFdp, kStreamtypes);
+ config.offload_info.sync_id = mFdp.ConsumeIntegral<uint32_t>();
+ config.offload_info.usage = getValue(&mFdp, kUsages);
+
+ config.format = getValue(&mFdp, kFormats);
+ config.sample_rate = mFdp.ConsumeIntegral<uint32_t>();
+
+ sp<DeviceDescriptorBase> device = new DeviceDescriptorBase(getValue(&mFdp, kDevices));
+ audio_output_flags_t flags = getValue(&mFdp, kOutputFlags);
+
+ media::OpenOutputRequest request{};
+ media::OpenOutputResponse response{};
+
+ request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
+ request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+ request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
+ request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
+
+ status_t status = af->openOutput(request, &response);
+ if (status != NO_ERROR) {
+ return NO_ERROR;
+ }
+ output = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_io_handle_t(response.output));
+
+ audio_io_handle_t output1 = mFdp.ConsumeIntegral<int32_t>();
+ af->openDuplicateOutput(output, output1);
+ af->suspendOutput(output);
+ af->restoreOutput(output);
+ af->closeOutput(output);
+ return NO_ERROR;
+}
+
+void AudioFlingerFuzzer::invokeAudioPatch() {
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (!af) {
+ return;
+ }
+ struct audio_patch patch = {};
+ audio_patch_handle_t handle = mFdp.ConsumeIntegral<int32_t>();
+
+ patch.id = mFdp.ConsumeIntegral<int32_t>();
+ patch.num_sources = mFdp.ConsumeIntegral<uint32_t>();
+ patch.num_sinks = mFdp.ConsumeIntegral<uint32_t>();
+
+ for (int i = 0; i < AUDIO_PATCH_PORTS_MAX; ++i) {
+ patch.sources[i].config_mask = mFdp.ConsumeIntegral<uint32_t>();
+ patch.sources[i].channel_mask = getValue(&mFdp, kChannelMasks);
+ patch.sources[i].format = getValue(&mFdp, kFormats);
+ patch.sources[i].gain.channel_mask = getValue(&mFdp, kChannelMasks);
+ patch.sources[i].gain.index = mFdp.ConsumeIntegral<int32_t>();
+ patch.sources[i].gain.mode = getValue(&mFdp, kGainModes);
+ patch.sources[i].gain.ramp_duration_ms = mFdp.ConsumeIntegral<uint32_t>();
+ patch.sources[i].id = static_cast<audio_format_t>(mFdp.ConsumeIntegral<int32_t>());
+ patch.sources[i].role = getValue(&mFdp, kPortRoles);
+ patch.sources[i].sample_rate = mFdp.ConsumeIntegral<uint32_t>();
+ patch.sources[i].type = getValue(&mFdp, kPortTypes);
+
+ patch.sinks[i].config_mask = mFdp.ConsumeIntegral<uint32_t>();
+ patch.sinks[i].channel_mask = getValue(&mFdp, kChannelMasks);
+ patch.sinks[i].format = getValue(&mFdp, kFormats);
+ patch.sinks[i].gain.channel_mask = getValue(&mFdp, kChannelMasks);
+ patch.sinks[i].gain.index = mFdp.ConsumeIntegral<int32_t>();
+ patch.sinks[i].gain.mode = getValue(&mFdp, kGainModes);
+ patch.sinks[i].gain.ramp_duration_ms = mFdp.ConsumeIntegral<uint32_t>();
+ patch.sinks[i].id = static_cast<audio_format_t>(mFdp.ConsumeIntegral<int32_t>());
+ patch.sinks[i].role = getValue(&mFdp, kPortRoles);
+ patch.sinks[i].sample_rate = mFdp.ConsumeIntegral<uint32_t>();
+ patch.sinks[i].type = getValue(&mFdp, kPortTypes);
+ }
+
+ status_t status = af->createAudioPatch(&patch, &handle);
+ if (status != NO_ERROR) {
+ return;
+ }
+
+ unsigned int num_patches = mFdp.ConsumeIntegral<uint32_t>();
+ struct audio_patch patches = {};
+ af->listAudioPatches(&num_patches, &patches);
+ af->releaseAudioPatch(handle);
+}
+
+void AudioFlingerFuzzer::process() {
+ invokeAudioEffect();
+ invokeAudioInputDevice();
+ invokeAudioOutputDevice();
+ invokeAudioPatch();
+ invokeAudioRecord();
+ invokeAudioSystem();
+ invokeAudioTrack();
+}
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {
+ if (size < 1) {
+ return 0;
+ }
+ AudioFlingerFuzzer audioFuzzer(data, size);
+ audioFuzzer.process();
+ return 0;
+}
diff --git a/media/libstagefright/tests/Android.bp b/media/libstagefright/tests/Android.bp
index 5f3f72c..4a505d4 100644
--- a/media/libstagefright/tests/Android.bp
+++ b/media/libstagefright/tests/Android.bp
@@ -9,7 +9,6 @@
"libmedia",
"libstagefright",
"libstagefright_foundation",
- "libstagefright_omx",
"libutils",
"liblog",
],
@@ -17,11 +16,8 @@
include_dirs: [
"frameworks/av/media/libstagefright",
"frameworks/av/media/libstagefright/include",
- "frameworks/native/include/media/openmax",
],
- compile_multilib: "prefer32",
-
cflags: [
"-Werror",
"-Wall",
diff --git a/services/audiopolicy/fuzzer/Android.bp b/services/audiopolicy/fuzzer/Android.bp
new file mode 100644
index 0000000..21f6515
--- /dev/null
+++ b/services/audiopolicy/fuzzer/Android.bp
@@ -0,0 +1,55 @@
+/******************************************************************************
+ *
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ ******************************************************************************/
+
+cc_fuzz {
+ name: "audiopolicy_fuzzer",
+ srcs: [
+ "audiopolicy_fuzzer.cpp",
+ ],
+ include_dirs: [
+ "frameworks/av/services/audiopolicy",
+ ],
+ shared_libs: [
+ "android.hardware.audio.common-util",
+ "capture_state_listener-aidl-cpp",
+ "libaudioclient",
+ "libaudiofoundation",
+ "libbase",
+ "libcutils",
+ "libhidlbase",
+ "libdl",
+ "liblog",
+ "libmedia_helper",
+ "libmediametrics",
+ "libutils",
+ "libxml2",
+ "libbinder",
+ "libaudiopolicy",
+ "libaudiopolicymanagerdefault",
+ ],
+ static_libs: [
+ "android.hardware.audio.common@7.0-enums",
+ "libaudiopolicycomponents",
+ ],
+ header_libs: [
+ "libaudiopolicycommon",
+ "libaudiopolicyengine_interface_headers",
+ "libaudiopolicymanager_interface_headers",
+ ],
+ data: [":audiopolicyfuzzer_configuration_files"],
+}
diff --git a/services/audiopolicy/fuzzer/README.md b/services/audiopolicy/fuzzer/README.md
new file mode 100644
index 0000000..08d7213
--- /dev/null
+++ b/services/audiopolicy/fuzzer/README.md
@@ -0,0 +1,63 @@
+# Fuzzer for libaudiopolicy
+
+## Plugin Design Considerations
+The fuzzer plugin for libaudiopolicy is designed based on the
+understanding of the service and tries to achieve the following:
+
+##### Maximize code coverage
+The configuration parameters are not hardcoded, but instead selected based on
+incoming data. This ensures more code paths are reached by the fuzzer.
+
+AudioPolicy APIs contain the following parameters:
+1. AudioFormats
+2. AudioChannelMasks
+3. AudioOutputFlags
+4. AudioDevices
+5. MixTypes
+6. MixRouteFlags
+7. SampleRates
+8. AudioUsages
+9. AudioContentTypes
+10. AudioSources
+11. AudioFlagMasks
+12. AudioPolicyDeviceStates
+
+| Parameter| Valid Input Values| Configured Value|
+|------------- |-------------| ----- |
+| `AudioFormat` | 77 values of type `audio_format_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `AudioChannelMask` | 83 values of type `audio_channel_mask_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `AudioOutputFlag` | 16 values of type `audio_output_flags_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `AudioDevice` | `AUDIO_DEVICE_OUT_AUX_DIGITAL`, `AUDIO_DEVICE_OUT_STUB`, `AUDIO_DEVICE_IN_VOICE_CALL`, `AUDIO_DEVICE_IN_AUX_DIGITAL`, `AUDIO_DEVICE_IN_STUB` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `MixType` | `MIX_TYPE_PLAYERS`, `MIX_TYPE_RECORDERS` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `MixRouteFlag` | `MIX_ROUTE_FLAG_RENDER`, `MIX_ROUTE_FLAG_LOOP_BACK`, `MIX_ROUTE_FLAG_LOOP_BACK_AND_RENDER`, `MIX_ROUTE_FLAG_ALL` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `SampleRate` | `0` to `UINT32_MAX` | Value obtained from FuzzedDataProvider |
+| `AudioUsage` | `AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST`, `AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT`, `AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED`, `AUDIO_USAGE_NOTIFICATION_EVENT` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `AudioContentType` | `AUDIO_CONTENT_TYPE_UNKNOWN`, `AUDIO_CONTENT_TYPE_SPEECH`, `AUDIO_CONTENT_TYPE_MUSIC`, `AUDIO_CONTENT_TYPE_MOVIE`, `AUDIO_CONTENT_TYPE_SONIFICATION` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `AudioSource` | 14 values of type `audio_source_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `AudioFlagMask` | 15 values of type `audio_flags_mask_t` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+| `AudioPolicyDeviceStates` | `AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE`, `AUDIO_POLICY_DEVICE_STATE_AVAILABLE`, `AUDIO_POLICY_DEVICE_STATE_CNT` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
+
+This also ensures that the plugin is always deterministic for any given input.
+
+## Build
+
+This describes steps to build audiopolicy_fuzzer binary.
+
+### Android
+
+#### Steps to build
+Build the fuzzer
+```
+ $ mm -j$(nproc) audiopolicy_fuzzer
+```
+
+#### Steps to run
+To run on device
+```
+ $ adb sync data
+ $ adb shell /data/fuzz/arm64/audiopolicy_fuzzer/audiopolicy_fuzzer
+```
+
+## References:
+ * http://llvm.org/docs/LibFuzzer.html
+ * https://github.com/google/oss-fuzz
diff --git a/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
new file mode 100644
index 0000000..c1f2aa8
--- /dev/null
+++ b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
@@ -0,0 +1,972 @@
+/******************************************************************************
+ *
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ ******************************************************************************/
+#include <stdint.h>
+#include <sys/wait.h>
+#include <unistd.h>
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include <Serializer.h>
+#include <android-base/file.h>
+#include <libxml/parser.h>
+#include <libxml/xinclude.h>
+#include <media/AudioPolicy.h>
+#include <media/PatchBuilder.h>
+#include <media/RecordingActivityTracker.h>
+
+#include <AudioPolicyInterface.h>
+#include <android_audio_policy_configuration_V7_0-enums.h>
+#include <fuzzer/FuzzedDataProvider.h>
+#include <tests/AudioPolicyManagerTestClient.h>
+#include <tests/AudioPolicyTestClient.h>
+#include <tests/AudioPolicyTestManager.h>
+#include <xsdc/XsdcSupport.h>
+
+using namespace android;
+
+namespace xsd {
+using namespace ::android::audio::policy::configuration::V7_0;
+}
+
+static const std::vector<audio_format_t> kAudioFormats = [] {
+ std::vector<audio_format_t> result;
+ for (const auto enumVal : xsdc_enum_range<xsd::AudioFormat>{}) {
+ audio_format_t audioFormatHal;
+ std::string audioFormat = toString(enumVal);
+ if (audio_format_from_string(audioFormat.c_str(), &audioFormatHal)) {
+ result.push_back(audioFormatHal);
+ }
+ }
+ return result;
+}();
+
+static const std::vector<audio_channel_mask_t> kAudioChannelOutMasks = [] {
+ std::vector<audio_channel_mask_t> result;
+ for (const auto enumVal : xsdc_enum_range<xsd::AudioChannelMask>{}) {
+ audio_channel_mask_t audioChannelMaskHal;
+ std::string audioChannelMask = toString(enumVal);
+ if (enumVal != xsd::AudioChannelMask::AUDIO_CHANNEL_NONE &&
+ audioChannelMask.find("_IN_") == std::string::npos &&
+ audio_channel_mask_from_string(audioChannelMask.c_str(), &audioChannelMaskHal)) {
+ result.push_back(audioChannelMaskHal);
+ }
+ }
+ return result;
+}();
+
+static const std::vector<audio_channel_mask_t> kAudioChannelInMasks = [] {
+ std::vector<audio_channel_mask_t> result;
+ for (const auto enumVal : xsdc_enum_range<xsd::AudioChannelMask>{}) {
+ audio_channel_mask_t audioChannelMaskHal;
+ std::string audioChannelMask = toString(enumVal);
+ if (enumVal != xsd::AudioChannelMask::AUDIO_CHANNEL_NONE &&
+ audioChannelMask.find("_OUT_") == std::string::npos &&
+ audio_channel_mask_from_string(audioChannelMask.c_str(), &audioChannelMaskHal)) {
+ result.push_back(audioChannelMaskHal);
+ }
+ }
+ return result;
+}();
+
+static const std::vector<audio_output_flags_t> kAudioOutputFlags = [] {
+ std::vector<audio_output_flags_t> result;
+ for (const auto enumVal : xsdc_enum_range<xsd::AudioInOutFlag>{}) {
+ audio_output_flags_t audioOutputFlagHal;
+ std::string audioOutputFlag = toString(enumVal);
+ if (audioOutputFlag.find("_OUTPUT_") != std::string::npos &&
+ audio_output_flag_from_string(audioOutputFlag.c_str(), &audioOutputFlagHal)) {
+ result.push_back(audioOutputFlagHal);
+ }
+ }
+ return result;
+}();
+
+static const std::vector<audio_devices_t> kAudioDevices = [] {
+ std::vector<audio_devices_t> result;
+ for (const auto enumVal : xsdc_enum_range<xsd::AudioDevice>{}) {
+ audio_devices_t audioDeviceHal;
+ std::string audioDevice = toString(enumVal);
+ if (audio_device_from_string(audioDevice.c_str(), &audioDeviceHal)) {
+ result.push_back(audioDeviceHal);
+ }
+ }
+ return result;
+}();
+
+static const std::vector<audio_usage_t> kAudioUsages = [] {
+ std::vector<audio_usage_t> result;
+ for (const auto enumVal : xsdc_enum_range<xsd::AudioUsage>{}) {
+ audio_usage_t audioUsageHal;
+ std::string audioUsage = toString(enumVal);
+ if (audio_usage_from_string(audioUsage.c_str(), &audioUsageHal)) {
+ result.push_back(audioUsageHal);
+ }
+ }
+ return result;
+}();
+
+static const std::vector<audio_source_t> kAudioSources = [] {
+ std::vector<audio_source_t> result;
+ for (const auto enumVal : xsdc_enum_range<xsd::AudioSource>{}) {
+ audio_source_t audioSourceHal;
+ std::string audioSource = toString(enumVal);
+ if (audio_source_from_string(audioSource.c_str(), &audioSourceHal)) {
+ result.push_back(audioSourceHal);
+ }
+ }
+ return result;
+}();
+
+static const std::vector<audio_content_type_t> kAudioContentTypes = [] {
+ std::vector<audio_content_type_t> result;
+ for (const auto enumVal : xsdc_enum_range<xsd::AudioContentType>{}) {
+ audio_content_type_t audioContentTypeHal;
+ std::string audioContentType = toString(enumVal);
+ if (audio_content_type_from_string(audioContentType.c_str(), &audioContentTypeHal)) {
+ result.push_back(audioContentTypeHal);
+ }
+ }
+ return result;
+}();
+
+std::vector<int> kMixTypes = {MIX_TYPE_PLAYERS, MIX_TYPE_RECORDERS};
+
+std::vector<int> kMixRouteFlags = {MIX_ROUTE_FLAG_RENDER, MIX_ROUTE_FLAG_LOOP_BACK,
+ MIX_ROUTE_FLAG_LOOP_BACK_AND_RENDER, MIX_ROUTE_FLAG_ALL};
+
+std::vector<audio_flags_mask_t> kAudioFlagMasks = {
+ AUDIO_FLAG_NONE, AUDIO_FLAG_AUDIBILITY_ENFORCED,
+ AUDIO_FLAG_SECURE, AUDIO_FLAG_SCO,
+ AUDIO_FLAG_BEACON, AUDIO_FLAG_HW_AV_SYNC,
+ AUDIO_FLAG_HW_HOTWORD, AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY,
+ AUDIO_FLAG_BYPASS_MUTE, AUDIO_FLAG_LOW_LATENCY,
+ AUDIO_FLAG_DEEP_BUFFER, AUDIO_FLAG_NO_MEDIA_PROJECTION,
+ AUDIO_FLAG_MUTE_HAPTIC, AUDIO_FLAG_NO_SYSTEM_CAPTURE,
+ AUDIO_FLAG_CAPTURE_PRIVATE};
+
+std::vector<audio_policy_dev_state_t> kAudioPolicyDeviceStates = {
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ AUDIO_POLICY_DEVICE_STATE_CNT,
+};
+
+std::vector<uint32_t> kSamplingRates = {8000, 16000, 44100, 48000, 88200, 96000};
+
+template <typename T>
+T getValueFromVector(FuzzedDataProvider *fdp, std::vector<T> arr) {
+ if (fdp->ConsumeBool()) {
+ return arr[fdp->ConsumeIntegralInRange<int32_t>(0, arr.size() - 1)];
+ } else {
+ return (T)fdp->ConsumeIntegral<uint32_t>();
+ }
+}
+
+class AudioPolicyManagerFuzzer {
+ public:
+ explicit AudioPolicyManagerFuzzer(FuzzedDataProvider *fdp);
+ virtual ~AudioPolicyManagerFuzzer() = default;
+ virtual bool initialize();
+ virtual void SetUpManagerConfig();
+ bool getOutputForAttr(audio_port_handle_t *selectedDeviceId, audio_format_t format,
+ audio_channel_mask_t channelMask, int sampleRate,
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ audio_io_handle_t *output = nullptr,
+ audio_port_handle_t *portId = nullptr, audio_attributes_t attr = {});
+ bool getInputForAttr(const audio_attributes_t &attr, audio_unique_id_t riid,
+ audio_port_handle_t *selectedDeviceId, audio_format_t format,
+ audio_channel_mask_t channelMask, int sampleRate,
+ audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
+ audio_port_handle_t *portId = nullptr);
+ bool findDevicePort(audio_port_role_t role, audio_devices_t deviceType,
+ const std::string &address, audio_port_v7 *foundPort);
+ static audio_port_handle_t getDeviceIdFromPatch(const struct audio_patch *patch);
+ audio_patch createFuzzedPatch();
+ void fuzzPatchCreation();
+ virtual void process();
+
+ protected:
+ std::unique_ptr<AudioPolicyManagerTestClient> mClient{new AudioPolicyManagerTestClient};
+ std::unique_ptr<AudioPolicyTestManager> mManager{new AudioPolicyTestManager(mClient.get())};
+ FuzzedDataProvider *mFdp;
+};
+
+AudioPolicyManagerFuzzer::AudioPolicyManagerFuzzer(FuzzedDataProvider *fdp)
+ : mFdp(fdp) {}
+
+bool AudioPolicyManagerFuzzer::initialize() {
+ if (mFdp->remaining_bytes() < 1) {
+ return false;
+ }
+ // init code
+ SetUpManagerConfig();
+
+ if (mManager->initialize() != NO_ERROR) {
+ return false;
+ }
+ if (mManager->initCheck() != NO_ERROR) {
+ return false;
+ }
+ return true;
+}
+
+void AudioPolicyManagerFuzzer::SetUpManagerConfig() { mManager->getConfig().setDefault(); }
+
+bool AudioPolicyManagerFuzzer::getOutputForAttr(
+ audio_port_handle_t *selectedDeviceId, audio_format_t format, audio_channel_mask_t channelMask,
+ int sampleRate, audio_output_flags_t flags, audio_io_handle_t *output,
+ audio_port_handle_t *portId, audio_attributes_t attr) {
+ audio_io_handle_t localOutput;
+ if (!output) output = &localOutput;
+ *output = AUDIO_IO_HANDLE_NONE;
+ audio_stream_type_t stream = AUDIO_STREAM_DEFAULT;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = sampleRate;
+ config.channel_mask = channelMask;
+ config.format = format;
+ audio_port_handle_t localPortId;
+ if (!portId) portId = &localPortId;
+ *portId = AUDIO_PORT_HANDLE_NONE;
+ AudioPolicyInterface::output_type_t outputType;
+
+ if (mManager->getOutputForAttr(&attr, output, AUDIO_SESSION_NONE, &stream, 0 /*uid*/, &config,
+ &flags, selectedDeviceId, portId, {}, &outputType) != OK) {
+ return false;
+ }
+ if (*output == AUDIO_IO_HANDLE_NONE || *portId == AUDIO_PORT_HANDLE_NONE) {
+ return false;
+ }
+ return true;
+}
+
+bool AudioPolicyManagerFuzzer::getInputForAttr(
+ const audio_attributes_t &attr, audio_unique_id_t riid, audio_port_handle_t *selectedDeviceId,
+ audio_format_t format, audio_channel_mask_t channelMask, int sampleRate,
+ audio_input_flags_t flags, audio_port_handle_t *portId) {
+ audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
+ audio_config_base_t config = AUDIO_CONFIG_BASE_INITIALIZER;
+ config.sample_rate = sampleRate;
+ config.channel_mask = channelMask;
+ config.format = format;
+ audio_port_handle_t localPortId;
+ if (!portId) portId = &localPortId;
+ *portId = AUDIO_PORT_HANDLE_NONE;
+ AudioPolicyInterface::input_type_t inputType;
+
+ if (mManager->getInputForAttr(&attr, &input, riid, AUDIO_SESSION_NONE, 0 /*uid*/, &config,
+ flags, selectedDeviceId, &inputType, portId) != OK) {
+ return false;
+ }
+ if (*portId == AUDIO_PORT_HANDLE_NONE || input == AUDIO_IO_HANDLE_NONE) {
+ return false;
+ }
+ return true;
+}
+
+bool AudioPolicyManagerFuzzer::findDevicePort(audio_port_role_t role, audio_devices_t deviceType,
+ const std::string &address,
+ audio_port_v7 *foundPort) {
+ uint32_t numPorts = 0;
+ uint32_t generation1;
+ status_t ret;
+
+ ret = mManager->listAudioPorts(role, AUDIO_PORT_TYPE_DEVICE, &numPorts, nullptr, &generation1);
+ if (ret != NO_ERROR) {
+ return false;
+ }
+
+ uint32_t generation2;
+ struct audio_port_v7 ports[numPorts];
+ ret = mManager->listAudioPorts(role, AUDIO_PORT_TYPE_DEVICE, &numPorts, ports, &generation2);
+ if (ret != NO_ERROR) {
+ return false;
+ }
+
+ for (const auto &port : ports) {
+ if (port.role == role && port.ext.device.type == deviceType &&
+ (strncmp(port.ext.device.address, address.c_str(), AUDIO_DEVICE_MAX_ADDRESS_LEN) ==
+ 0)) {
+ if (foundPort) *foundPort = port;
+ return true;
+ }
+ }
+ return false;
+}
+
+audio_port_handle_t AudioPolicyManagerFuzzer::getDeviceIdFromPatch(
+ const struct audio_patch *patch) {
+ if (patch->num_sources != 0 && patch->num_sinks != 0) {
+ if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
+ return patch->sinks[0].id;
+ } else {
+ return patch->sources[0].id;
+ }
+ }
+ return AUDIO_PORT_HANDLE_NONE;
+}
+
+audio_patch AudioPolicyManagerFuzzer::createFuzzedPatch() {
+ audio_patch patch{};
+ patch.id = mFdp->ConsumeIntegral<uint32_t>();
+ patch.num_sources = mFdp->ConsumeIntegralInRange(0, AUDIO_PATCH_PORTS_MAX);
+ for (int i = 0; i < patch.num_sources; ++i) {
+ audio_port_config config{};
+ std::vector<uint8_t> bytes = mFdp->ConsumeBytes<uint8_t>(sizeof(config));
+ memcpy(reinterpret_cast<uint8_t *>(&config), &bytes[0], bytes.size());
+ patch.sources[i] = config;
+ }
+ patch.num_sinks = mFdp->ConsumeIntegralInRange(0, AUDIO_PATCH_PORTS_MAX);
+ for (int i = 0; i < patch.num_sinks; ++i) {
+ audio_port_config config{};
+ std::vector<uint8_t> bytes = mFdp->ConsumeBytes<uint8_t>(sizeof(config));
+ memcpy(reinterpret_cast<uint8_t *>(&config), &bytes[0], bytes.size());
+ patch.sinks[i] = config;
+ }
+ return patch;
+}
+
+void AudioPolicyManagerFuzzer::fuzzPatchCreation() {
+ if (mFdp->remaining_bytes()) {
+ audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
+ uid_t uid = mFdp->ConsumeIntegral<uint32_t>();
+
+ // create a fuzzed patch
+ handle = AUDIO_PATCH_HANDLE_NONE;
+ audio_patch patch = createFuzzedPatch();
+ uid = mFdp->ConsumeIntegral<uint32_t>();
+ if (mManager->createAudioPatch(&patch, &handle, uid) == NO_ERROR) {
+ mManager->releaseAudioPatch(handle, uid);
+ }
+ }
+}
+
+void AudioPolicyManagerFuzzer::process() {
+ if (initialize()) {
+ fuzzPatchCreation();
+ }
+}
+
+class AudioPolicyManagerFuzzerWithConfigurationFile : public AudioPolicyManagerFuzzer {
+ public:
+ explicit AudioPolicyManagerFuzzerWithConfigurationFile(FuzzedDataProvider *fdp)
+ : AudioPolicyManagerFuzzer(fdp){};
+
+ protected:
+ void SetUpManagerConfig() override;
+ virtual std::string getConfigFile();
+ void traverseAndFuzzXML(xmlDocPtr pDoc, xmlNodePtr curr);
+ std::string fuzzXML(std::string xmlPath);
+
+ static inline const std::string sExecutableDir = base::GetExecutableDirectory() + "/";
+ static inline const std::string sDefaultConfig =
+ sExecutableDir + "data/test_audio_policy_configuration.xml";
+ static inline const std::string sFuzzedConfig = sExecutableDir + "fuzzed.xml";;
+};
+
+std::string AudioPolicyManagerFuzzerWithConfigurationFile::getConfigFile() {
+ return fuzzXML(sDefaultConfig);
+}
+
+void AudioPolicyManagerFuzzerWithConfigurationFile::SetUpManagerConfig() {
+ deserializeAudioPolicyFile(getConfigFile().c_str(), &mManager->getConfig());
+}
+
+void AudioPolicyManagerFuzzerWithConfigurationFile::traverseAndFuzzXML(xmlDocPtr pDoc,
+ xmlNodePtr curr) {
+ if (curr == nullptr) {
+ return;
+ }
+
+ xmlAttr *attribute = curr->properties;
+ while (attribute) {
+ if (!xmlStrcmp(attribute->name, reinterpret_cast<const xmlChar *>("format"))) {
+ const char *newFormat =
+ audio_format_to_string(getValueFromVector<audio_format_t>(mFdp, kAudioFormats));
+ xmlSetProp(curr, attribute->name, reinterpret_cast<const xmlChar *>(newFormat));
+ }
+ if (!xmlStrcmp(attribute->name, reinterpret_cast<const xmlChar *>("flags"))) {
+ std::string newFlag = "";
+ uint16_t numFlags = std::max((uint16_t)1, mFdp->ConsumeIntegral<uint16_t>());
+ for (uint16_t i = 0; i < numFlags; ++i) {
+ newFlag += std::string(audio_output_flag_to_string(
+ getValueFromVector<audio_output_flags_t>(mFdp, kAudioOutputFlags)));
+ if (i != (numFlags - 1)) {
+ newFlag += std::string("|");
+ }
+ }
+ xmlSetProp(curr, attribute->name, reinterpret_cast<const xmlChar *>(newFlag.c_str()));
+ }
+ if (!xmlStrcmp(attribute->name, reinterpret_cast<const xmlChar *>("samplingRates"))) {
+ std::string newRate = "";
+ uint16_t numRates = std::max((uint16_t)1, mFdp->ConsumeIntegral<uint16_t>());
+ for (uint16_t i = 0; i < numRates; ++i) {
+ newRate += std::to_string(getValueFromVector<uint32_t>(mFdp, kSamplingRates));
+ if (i != (numRates - 1)) {
+ newRate += std::string(",");
+ }
+ }
+ xmlSetProp(curr, attribute->name, reinterpret_cast<const xmlChar *>(newRate.c_str()));
+ }
+ if (!xmlStrcmp(attribute->name, reinterpret_cast<const xmlChar *>("channelMasks"))) {
+ int isOutMask = -1;
+ char *value =
+ reinterpret_cast<char *>(xmlNodeListGetString(pDoc, attribute->children, 1));
+ if (std::string(value).find(std::string("_OUT_")) != std::string::npos) {
+ // OUT mask
+ isOutMask = 1;
+ } else if (std::string(value).find(std::string("_IN_")) != std::string::npos) {
+ // IN mask
+ isOutMask = 0;
+ }
+ if (isOutMask != -1) {
+ std::string newMask = "";
+ uint16_t numMasks = std::max((uint16_t)1, mFdp->ConsumeIntegral<uint16_t>());
+ for (uint16_t i = 0; i < numMasks; ++i) {
+ if (isOutMask) {
+ newMask += std::string(audio_channel_out_mask_to_string(
+ getValueFromVector<audio_channel_mask_t>(mFdp, kAudioChannelOutMasks)));
+ } else {
+ newMask += std::string(audio_channel_in_mask_to_string(
+ getValueFromVector<audio_channel_mask_t>(mFdp, kAudioChannelInMasks)));
+ }
+ if (i != (numMasks - 1)) {
+ newMask += std::string(",");
+ }
+ }
+ xmlSetProp(curr, attribute->name,
+ reinterpret_cast<const xmlChar *>(newMask.c_str()));
+ }
+ xmlFree(value);
+ }
+ attribute = attribute->next;
+ }
+
+ curr = curr->xmlChildrenNode;
+ while (curr != nullptr) {
+ traverseAndFuzzXML(pDoc, curr);
+ curr = curr->next;
+ }
+}
+
+std::string AudioPolicyManagerFuzzerWithConfigurationFile::fuzzXML(std::string xmlPath) {
+ std::string outPath = sFuzzedConfig;
+
+ // Load in the xml file from disk
+ xmlDocPtr pDoc = xmlParseFile(xmlPath.c_str());
+ xmlNodePtr root = xmlDocGetRootElement(pDoc);
+
+ traverseAndFuzzXML(pDoc, root);
+
+ // Save the document back out to disk.
+ xmlSaveFileEnc(outPath.c_str(), pDoc, "UTF-8");
+ xmlFreeDoc(pDoc);
+
+ return outPath;
+}
+
+class AudioPolicyManagerFuzzerMsd : public AudioPolicyManagerFuzzerWithConfigurationFile {
+ public:
+ explicit AudioPolicyManagerFuzzerMsd(FuzzedDataProvider *fdp)
+ : AudioPolicyManagerFuzzerWithConfigurationFile(fdp) {}
+
+ protected:
+ std::string getConfigFile() override;
+
+ static inline const std::string sMsdConfig =
+ sExecutableDir + "data/test_audio_policy_msd_configuration.xml";
+};
+
+std::string AudioPolicyManagerFuzzerMsd::getConfigFile() { return fuzzXML(sMsdConfig); }
+
+using PolicyMixTuple = std::tuple<audio_usage_t, audio_source_t, uint32_t>;
+
+class AudioPolicyManagerFuzzerDynamicPolicy : public AudioPolicyManagerFuzzerWithConfigurationFile {
+ public:
+ explicit AudioPolicyManagerFuzzerDynamicPolicy(FuzzedDataProvider *fdp)
+ : AudioPolicyManagerFuzzerWithConfigurationFile(fdp){};
+ ~AudioPolicyManagerFuzzerDynamicPolicy() override;
+ void process() override;
+
+ protected:
+ status_t addPolicyMix(int mixType, int mixFlag, audio_devices_t deviceType,
+ std::string mixAddress, const audio_config_t &audioConfig,
+ const std::vector<PolicyMixTuple> &rules);
+ void clearPolicyMix();
+ void registerPolicyMixes();
+ void unregisterPolicyMixes();
+
+ Vector<AudioMix> mAudioMixes;
+ const std::string mMixAddress = "remote_submix_media";
+};
+
+AudioPolicyManagerFuzzerDynamicPolicy::~AudioPolicyManagerFuzzerDynamicPolicy() {
+ clearPolicyMix();
+}
+
+status_t AudioPolicyManagerFuzzerDynamicPolicy::addPolicyMix(
+ int mixType, int mixFlag, audio_devices_t deviceType, std::string mixAddress,
+ const audio_config_t &audioConfig, const std::vector<PolicyMixTuple> &rules) {
+ Vector<AudioMixMatchCriterion> myMixMatchCriteria;
+
+ for (const auto &rule : rules) {
+ myMixMatchCriteria.add(
+ AudioMixMatchCriterion(std::get<0>(rule), std::get<1>(rule), std::get<2>(rule)));
+ }
+
+ AudioMix myAudioMix(myMixMatchCriteria, mixType, audioConfig, mixFlag,
+ String8(mixAddress.c_str()), 0);
+ myAudioMix.mDeviceType = deviceType;
+ // Clear mAudioMix before add new one to make sure we don't add already existing mixes.
+ mAudioMixes.clear();
+ mAudioMixes.add(myAudioMix);
+
+ // As the policy mixes registration may fail at some case,
+ // caller need to check the returned status.
+ status_t ret = mManager->registerPolicyMixes(mAudioMixes);
+ return ret;
+}
+
+void AudioPolicyManagerFuzzerDynamicPolicy::clearPolicyMix() {
+ if (mManager != nullptr) {
+ mManager->unregisterPolicyMixes(mAudioMixes);
+ }
+ mAudioMixes.clear();
+}
+
+void AudioPolicyManagerFuzzerDynamicPolicy::registerPolicyMixes() {
+ const uint32_t numPolicies = mFdp->ConsumeIntegralInRange<uint32_t>(1, MAX_MIXES_PER_POLICY);
+
+ for (int i = 0; i < numPolicies; ++i) {
+ audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+ audioConfig.channel_mask = getValueFromVector<audio_channel_mask_t>(
+ mFdp, mFdp->ConsumeBool() ? kAudioChannelInMasks : kAudioChannelOutMasks);
+ audioConfig.format = getValueFromVector<audio_format_t>(mFdp, kAudioFormats);
+ audioConfig.sample_rate = getValueFromVector<uint32_t>(mFdp, kSamplingRates);
+ addPolicyMix(getValueFromVector<int>(mFdp, kMixTypes),
+ getValueFromVector<int>(mFdp, kMixRouteFlags),
+ getValueFromVector<audio_devices_t>(mFdp, kAudioDevices), "", audioConfig,
+ std::vector<PolicyMixTuple>());
+ }
+}
+
+void AudioPolicyManagerFuzzerDynamicPolicy::unregisterPolicyMixes() {
+ mManager->unregisterPolicyMixes(mAudioMixes);
+}
+
+void AudioPolicyManagerFuzzerDynamicPolicy::process() {
+ if (initialize()) {
+ registerPolicyMixes();
+ fuzzPatchCreation();
+ unregisterPolicyMixes();
+ }
+}
+
+class AudioPolicyManagerFuzzerDPNoRemoteSubmixModule
+ : public AudioPolicyManagerFuzzerDynamicPolicy {
+ public:
+ explicit AudioPolicyManagerFuzzerDPNoRemoteSubmixModule(FuzzedDataProvider *fdp)
+ : AudioPolicyManagerFuzzerDynamicPolicy(fdp){};
+
+ protected:
+ std::string getConfigFile() override;
+
+ static inline const std::string sPrimaryOnlyConfig =
+ sExecutableDir + "data/test_audio_policy_primary_only_configuration.xml";
+};
+
+std::string AudioPolicyManagerFuzzerDPNoRemoteSubmixModule::getConfigFile() {
+ return fuzzXML(sPrimaryOnlyConfig);
+}
+
+class AudioPolicyManagerFuzzerDPPlaybackReRouting : public AudioPolicyManagerFuzzerDynamicPolicy {
+ public:
+ explicit AudioPolicyManagerFuzzerDPPlaybackReRouting(FuzzedDataProvider *fdp);
+ ~AudioPolicyManagerFuzzerDPPlaybackReRouting() override;
+ void process() override;
+
+ protected:
+ bool initialize() override;
+ void playBackReRouting();
+
+ std::unique_ptr<RecordingActivityTracker> mTracker;
+
+ std::vector<PolicyMixTuple> mUsageRules = {
+ {AUDIO_USAGE_MEDIA, AUDIO_SOURCE_DEFAULT, RULE_MATCH_ATTRIBUTE_USAGE},
+ {AUDIO_USAGE_ALARM, AUDIO_SOURCE_DEFAULT, RULE_MATCH_ATTRIBUTE_USAGE}};
+
+ struct audio_port_v7 mInjectionPort;
+ audio_port_handle_t mPortId = AUDIO_PORT_HANDLE_NONE;
+ audio_config_t mAudioConfig;
+};
+
+AudioPolicyManagerFuzzerDPPlaybackReRouting::AudioPolicyManagerFuzzerDPPlaybackReRouting(
+ FuzzedDataProvider *fdp)
+ : AudioPolicyManagerFuzzerDynamicPolicy(fdp) {
+ const uint32_t numRules = mFdp->ConsumeIntegralInRange<uint32_t>(1, 10);
+ for (int i = 0; i < numRules; ++i) {
+ PolicyMixTuple rule = {getValueFromVector<audio_usage_t>(mFdp, kAudioUsages),
+ getValueFromVector<audio_source_t>(mFdp, kAudioSources),
+ RULE_MATCH_ATTRIBUTE_USAGE};
+ mUsageRules.push_back(rule);
+ }
+}
+
+AudioPolicyManagerFuzzerDPPlaybackReRouting::~AudioPolicyManagerFuzzerDPPlaybackReRouting() {
+ mManager->stopInput(mPortId);
+}
+
+bool AudioPolicyManagerFuzzerDPPlaybackReRouting::initialize() {
+ AudioPolicyManagerFuzzerDynamicPolicy::initialize();
+ mTracker.reset(new RecordingActivityTracker());
+
+ mAudioConfig = AUDIO_CONFIG_INITIALIZER;
+ mAudioConfig.channel_mask =
+ getValueFromVector<audio_channel_mask_t>(mFdp, kAudioChannelOutMasks);
+ mAudioConfig.format = getValueFromVector<audio_format_t>(mFdp, kAudioFormats);
+ mAudioConfig.sample_rate = getValueFromVector<uint32_t>(mFdp, kSamplingRates);
+ status_t ret = addPolicyMix(getValueFromVector<int>(mFdp, kMixTypes),
+ getValueFromVector<int>(mFdp, kMixRouteFlags),
+ getValueFromVector<audio_devices_t>(mFdp, kAudioDevices),
+ mMixAddress, mAudioConfig, mUsageRules);
+ if (ret != NO_ERROR) {
+ return false;
+ }
+
+ struct audio_port_v7 extractionPort;
+ findDevicePort(AUDIO_PORT_ROLE_SOURCE, getValueFromVector<audio_devices_t>(mFdp, kAudioDevices),
+ mMixAddress, &extractionPort);
+
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_source_t source = getValueFromVector<audio_source_t>(mFdp, kAudioSources);
+ audio_attributes_t attr = {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, source,
+ AUDIO_FLAG_NONE, ""};
+ std::string tags = "addr=" + mMixAddress;
+ strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
+ getInputForAttr(attr, mTracker->getRiid(), &selectedDeviceId, mAudioConfig.format,
+ mAudioConfig.channel_mask, mAudioConfig.sample_rate, AUDIO_INPUT_FLAG_NONE,
+ &mPortId);
+
+ ret = mManager->startInput(mPortId);
+ if (ret != NO_ERROR) {
+ return false;
+ }
+ if (!findDevicePort(AUDIO_PORT_ROLE_SINK,
+ getValueFromVector<audio_devices_t>(mFdp, kAudioDevices), mMixAddress,
+ &mInjectionPort)) {
+ return false;
+ }
+
+ return true;
+}
+
+void AudioPolicyManagerFuzzerDPPlaybackReRouting::playBackReRouting() {
+ const uint32_t numTestCases = mFdp->ConsumeIntegralInRange<uint32_t>(1, 10);
+ for (int i = 0; i < numTestCases; ++i) {
+ audio_attributes_t attr;
+ attr.content_type = getValueFromVector<audio_content_type_t>(mFdp, kAudioContentTypes);
+ attr.usage = getValueFromVector<audio_usage_t>(mFdp, kAudioUsages);
+ attr.source = getValueFromVector<audio_source_t>(mFdp, kAudioSources);
+ attr.flags = getValueFromVector<audio_flags_mask_t>(mFdp, kAudioFlagMasks);
+ std::string tags(mFdp->ConsumeBool() ? "" : "addr=remote_submix_media");
+ strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
+
+ audio_port_handle_t playbackRoutedPortId = AUDIO_PORT_HANDLE_NONE;
+ getOutputForAttr(&playbackRoutedPortId, mAudioConfig.format, mAudioConfig.channel_mask,
+ mAudioConfig.sample_rate, AUDIO_OUTPUT_FLAG_NONE, nullptr /*output*/,
+ nullptr /*portId*/, attr);
+ }
+}
+
+void AudioPolicyManagerFuzzerDPPlaybackReRouting::process() {
+ if (initialize()) {
+ playBackReRouting();
+ registerPolicyMixes();
+ fuzzPatchCreation();
+ unregisterPolicyMixes();
+ }
+}
+
+class AudioPolicyManagerFuzzerDPMixRecordInjection : public AudioPolicyManagerFuzzerDynamicPolicy {
+ public:
+ explicit AudioPolicyManagerFuzzerDPMixRecordInjection(FuzzedDataProvider *fdp);
+ ~AudioPolicyManagerFuzzerDPMixRecordInjection() override;
+ void process() override;
+
+ protected:
+ bool initialize() override;
+ void recordingInjection();
+
+ std::unique_ptr<RecordingActivityTracker> mTracker;
+
+ std::vector<PolicyMixTuple> mSourceRules = {
+ {AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_CAMCORDER, RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET},
+ {AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_MIC, RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET},
+ {AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_VOICE_COMMUNICATION,
+ RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET}};
+
+ struct audio_port_v7 mExtractionPort;
+ audio_port_handle_t mPortId = AUDIO_PORT_HANDLE_NONE;
+ audio_config_t mAudioConfig;
+};
+
+AudioPolicyManagerFuzzerDPMixRecordInjection::AudioPolicyManagerFuzzerDPMixRecordInjection(
+ FuzzedDataProvider *fdp)
+ : AudioPolicyManagerFuzzerDynamicPolicy(fdp) {
+ const uint32_t numRules = mFdp->ConsumeIntegralInRange<uint32_t>(1, 10);
+ for (int i = 0; i < numRules; ++i) {
+ PolicyMixTuple rule = {getValueFromVector<audio_usage_t>(mFdp, kAudioUsages),
+ getValueFromVector<audio_source_t>(mFdp, kAudioSources),
+ RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET};
+ mSourceRules.push_back(rule);
+ }
+}
+
+AudioPolicyManagerFuzzerDPMixRecordInjection::~AudioPolicyManagerFuzzerDPMixRecordInjection() {
+ mManager->stopOutput(mPortId);
+}
+
+bool AudioPolicyManagerFuzzerDPMixRecordInjection::initialize() {
+ AudioPolicyManagerFuzzerDynamicPolicy::initialize();
+
+ mTracker.reset(new RecordingActivityTracker());
+
+ mAudioConfig = AUDIO_CONFIG_INITIALIZER;
+ mAudioConfig.channel_mask =
+ getValueFromVector<audio_channel_mask_t>(mFdp, kAudioChannelInMasks);
+ mAudioConfig.format = getValueFromVector<audio_format_t>(mFdp, kAudioFormats);
+ mAudioConfig.sample_rate = getValueFromVector<uint32_t>(mFdp, kSamplingRates);
+ status_t ret = addPolicyMix(getValueFromVector<int>(mFdp, kMixTypes),
+ getValueFromVector<int>(mFdp, kMixRouteFlags),
+ getValueFromVector<audio_devices_t>(mFdp, kAudioDevices),
+ mMixAddress, mAudioConfig, mSourceRules);
+ if (ret != NO_ERROR) {
+ return false;
+ }
+
+ struct audio_port_v7 injectionPort;
+ findDevicePort(AUDIO_PORT_ROLE_SINK, getValueFromVector<audio_devices_t>(mFdp, kAudioDevices),
+ mMixAddress, &injectionPort);
+
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_usage_t usage = getValueFromVector<audio_usage_t>(mFdp, kAudioUsages);
+ audio_attributes_t attr = {AUDIO_CONTENT_TYPE_UNKNOWN, usage, AUDIO_SOURCE_DEFAULT,
+ AUDIO_FLAG_NONE, ""};
+ std::string tags = std::string("addr=") + mMixAddress;
+ strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
+ getOutputForAttr(&selectedDeviceId, mAudioConfig.format, mAudioConfig.channel_mask,
+ mAudioConfig.sample_rate /*sampleRate*/, AUDIO_OUTPUT_FLAG_NONE,
+ nullptr /*output*/, &mPortId, attr);
+ ret = mManager->startOutput(mPortId);
+ if (ret != NO_ERROR) {
+ return false;
+ }
+ getDeviceIdFromPatch(mClient->getLastAddedPatch());
+ if (!findDevicePort(AUDIO_PORT_ROLE_SOURCE,
+ getValueFromVector<audio_devices_t>(mFdp, kAudioDevices), mMixAddress,
+ &mExtractionPort)) {
+ return false;
+ }
+
+ return true;
+}
+
+void AudioPolicyManagerFuzzerDPMixRecordInjection::recordingInjection() {
+ const uint32_t numTestCases = mFdp->ConsumeIntegralInRange<uint32_t>(1, 10);
+ for (int i = 0; i < numTestCases; ++i) {
+ audio_attributes_t attr;
+ attr.content_type = getValueFromVector<audio_content_type_t>(mFdp, kAudioContentTypes);
+ attr.usage = getValueFromVector<audio_usage_t>(mFdp, kAudioUsages);
+ attr.source = getValueFromVector<audio_source_t>(mFdp, kAudioSources);
+ attr.flags = getValueFromVector<audio_flags_mask_t>(mFdp, kAudioFlagMasks);
+ std::string tags(mFdp->ConsumeBool() ? "" : "addr=remote_submix_media");
+ strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
+
+ audio_port_handle_t captureRoutedPortId = AUDIO_PORT_HANDLE_NONE;
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+ getInputForAttr(attr, mTracker->getRiid(), &captureRoutedPortId, mAudioConfig.format,
+ mAudioConfig.channel_mask, mAudioConfig.sample_rate, AUDIO_INPUT_FLAG_NONE,
+ &portId);
+ }
+}
+
+void AudioPolicyManagerFuzzerDPMixRecordInjection::process() {
+ if (initialize()) {
+ recordingInjection();
+ registerPolicyMixes();
+ fuzzPatchCreation();
+ unregisterPolicyMixes();
+ }
+}
+
+using DeviceConnectionTestParams =
+ std::tuple<audio_devices_t /*type*/, std::string /*name*/, std::string /*address*/>;
+
+class AudioPolicyManagerFuzzerDeviceConnection
+ : public AudioPolicyManagerFuzzerWithConfigurationFile {
+ public:
+ explicit AudioPolicyManagerFuzzerDeviceConnection(FuzzedDataProvider *fdp)
+ : AudioPolicyManagerFuzzerWithConfigurationFile(fdp){};
+ void process() override;
+
+ protected:
+ void setDeviceConnectionState();
+ void explicitlyRoutingAfterConnection();
+};
+
+void AudioPolicyManagerFuzzerDeviceConnection::setDeviceConnectionState() {
+ const uint32_t numTestCases = mFdp->ConsumeIntegralInRange<uint32_t>(1, 10);
+ for (int i = 0; i < numTestCases; ++i) {
+ const audio_devices_t type = getValueFromVector<audio_devices_t>(mFdp, kAudioDevices);
+ const std::string name = mFdp->ConsumeRandomLengthString();
+ const std::string address = mFdp->ConsumeRandomLengthString();
+ mManager->setDeviceConnectionState(
+ type, getValueFromVector<audio_policy_dev_state_t>(mFdp, kAudioPolicyDeviceStates),
+ address.c_str(), name.c_str(), getValueFromVector<audio_format_t>(mFdp, kAudioFormats));
+ }
+}
+
+void AudioPolicyManagerFuzzerDeviceConnection::explicitlyRoutingAfterConnection() {
+ const uint32_t numTestCases = mFdp->ConsumeIntegralInRange<uint32_t>(1, 10);
+ for (int i = 0; i < numTestCases; ++i) {
+ const audio_devices_t type = getValueFromVector<audio_devices_t>(mFdp, kAudioDevices);
+ const std::string name = mFdp->ConsumeRandomLengthString();
+ const std::string address = mFdp->ConsumeRandomLengthString();
+ mManager->setDeviceConnectionState(
+ type, getValueFromVector<audio_policy_dev_state_t>(mFdp, kAudioPolicyDeviceStates),
+ address.c_str(), name.c_str(), getValueFromVector<audio_format_t>(mFdp, kAudioFormats));
+
+ audio_port_v7 devicePort;
+ const audio_port_role_t role =
+ audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
+ findDevicePort(role, type, address, &devicePort);
+
+ audio_port_handle_t routedPortId = devicePort.id;
+ // Try start input or output according to the device type
+ if (audio_is_output_devices(type)) {
+ getOutputForAttr(&routedPortId, getValueFromVector<audio_format_t>(mFdp, kAudioFormats),
+ getValueFromVector<audio_channel_mask_t>(mFdp, kAudioChannelOutMasks),
+ getValueFromVector<uint32_t>(mFdp, kSamplingRates),
+ AUDIO_OUTPUT_FLAG_NONE);
+ } else if (audio_is_input_device(type)) {
+ RecordingActivityTracker tracker;
+ getInputForAttr({}, tracker.getRiid(), &routedPortId,
+ getValueFromVector<audio_format_t>(mFdp, kAudioFormats),
+ getValueFromVector<audio_channel_mask_t>(mFdp, kAudioChannelInMasks),
+ getValueFromVector<uint32_t>(mFdp, kSamplingRates),
+ AUDIO_INPUT_FLAG_NONE);
+ }
+ }
+}
+
+void AudioPolicyManagerFuzzerDeviceConnection::process() {
+ if (initialize()) {
+ setDeviceConnectionState();
+ explicitlyRoutingAfterConnection();
+ fuzzPatchCreation();
+ }
+}
+
+class AudioPolicyManagerTVFuzzer : public AudioPolicyManagerFuzzerWithConfigurationFile {
+ public:
+ explicit AudioPolicyManagerTVFuzzer(FuzzedDataProvider *fdp)
+ : AudioPolicyManagerFuzzerWithConfigurationFile(fdp){};
+ void process() override;
+
+ protected:
+ std::string getConfigFile();
+ void testHDMIPortSelection(audio_output_flags_t flags);
+
+ static inline const std::string sTvConfig =
+ AudioPolicyManagerTVFuzzer::sExecutableDir + "data/test_tv_apm_configuration.xml";
+};
+
+std::string AudioPolicyManagerTVFuzzer::getConfigFile() { return fuzzXML(sTvConfig); }
+
+void AudioPolicyManagerTVFuzzer::testHDMIPortSelection(audio_output_flags_t flags) {
+ audio_devices_t audioDevice = getValueFromVector<audio_devices_t>(mFdp, kAudioDevices);
+ audio_format_t audioFormat = getValueFromVector<audio_format_t>(mFdp, kAudioFormats);
+ status_t ret = mManager->setDeviceConnectionState(
+ audioDevice, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, "" /*address*/, "" /*name*/, audioFormat);
+ if (ret != NO_ERROR) {
+ return;
+ }
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_io_handle_t output;
+ audio_port_handle_t portId;
+ getOutputForAttr(&selectedDeviceId, getValueFromVector<audio_format_t>(mFdp, kAudioFormats),
+ getValueFromVector<audio_channel_mask_t>(mFdp, kAudioChannelOutMasks),
+ getValueFromVector<uint32_t>(mFdp, kSamplingRates), flags, &output, &portId);
+ sp<SwAudioOutputDescriptor> outDesc = mManager->getOutputs().valueFor(output);
+ if (outDesc.get() == nullptr) {
+ return;
+ }
+ audio_port_v7 port = {};
+ outDesc->toAudioPort(&port);
+ mManager->releaseOutput(portId);
+ mManager->setDeviceConnectionState(audioDevice, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ "" /*address*/, "" /*name*/, audioFormat);
+}
+
+void AudioPolicyManagerTVFuzzer::process() {
+ if (initialize()) {
+ testHDMIPortSelection(getValueFromVector<audio_output_flags_t>(mFdp, kAudioOutputFlags));
+ fuzzPatchCreation();
+ }
+}
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {
+ if (size < 1) {
+ return 0;
+ }
+ FuzzedDataProvider fdp = FuzzedDataProvider(data, size);
+ while (fdp.remaining_bytes() > 0) {
+ AudioPolicyManagerFuzzer audioPolicyManagerFuzzer(&fdp);
+ audioPolicyManagerFuzzer.process();
+
+ AudioPolicyManagerFuzzerMsd audioPolicyManagerFuzzerMsd(&fdp);
+ audioPolicyManagerFuzzerMsd.process();
+
+ AudioPolicyManagerFuzzerWithConfigurationFile audioPolicyManagerFuzzerWithConfigurationFile(
+ &fdp);
+ audioPolicyManagerFuzzerWithConfigurationFile.process();
+
+ AudioPolicyManagerFuzzerDynamicPolicy audioPolicyManagerFuzzerDynamicPolicy(&fdp);
+ audioPolicyManagerFuzzerDynamicPolicy.process();
+
+ AudioPolicyManagerFuzzerDPNoRemoteSubmixModule
+ audioPolicyManagerFuzzerDPNoRemoteSubmixModule(&fdp);
+ audioPolicyManagerFuzzerDPNoRemoteSubmixModule.process();
+
+ AudioPolicyManagerFuzzerDPPlaybackReRouting audioPolicyManagerFuzzerDPPlaybackReRouting(
+ &fdp);
+ audioPolicyManagerFuzzerDPPlaybackReRouting.process();
+
+ AudioPolicyManagerFuzzerDPMixRecordInjection audioPolicyManagerFuzzerDPMixRecordInjection(
+ &fdp);
+ audioPolicyManagerFuzzerDPMixRecordInjection.process();
+
+ AudioPolicyManagerFuzzerDeviceConnection audioPolicyManagerFuzzerDeviceConnection(&fdp);
+ audioPolicyManagerFuzzerDeviceConnection.process();
+
+ AudioPolicyManagerTVFuzzer audioPolicyManagerTVFuzzer(&fdp);
+ audioPolicyManagerTVFuzzer.process();
+ }
+ return 0;
+}
diff --git a/services/audiopolicy/fuzzer/resources/Android.bp b/services/audiopolicy/fuzzer/resources/Android.bp
new file mode 100644
index 0000000..f1e3a51
--- /dev/null
+++ b/services/audiopolicy/fuzzer/resources/Android.bp
@@ -0,0 +1,27 @@
+/******************************************************************************
+ *
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at:
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ ******************************************************************************/
+
+filegroup {
+ name: "audiopolicyfuzzer_configuration_files",
+ srcs: [
+ "test_audio_policy_configuration.xml",
+ "test_audio_policy_msd_configuration.xml",
+ "test_audio_policy_primary_only_configuration.xml",
+ "test_tv_apm_configuration.xml",
+ ],
+}
diff --git a/services/audiopolicy/fuzzer/resources/test_audio_policy_configuration.xml b/services/audiopolicy/fuzzer/resources/test_audio_policy_configuration.xml
new file mode 100644
index 0000000..7e26c33
--- /dev/null
+++ b/services/audiopolicy/fuzzer/resources/test_audio_policy_configuration.xml
@@ -0,0 +1,111 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2021 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <globalConfiguration speaker_drc_enabled="true"/>
+
+ <modules>
+ <!-- Primary module -->
+ <module name="primary" halVersion="2.0">
+ <attachedDevices>
+ <item>Speaker</item>
+ <item>Built-In Mic</item>
+ </attachedDevices>
+ <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <mixPorts>
+ <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="primary input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bt_hfp_output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bt_hfp_input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,11025,16000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+ </devicePort>
+ <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+ </devicePort>
+ <devicePort tagName="Hdmi" type="AUDIO_DEVICE_OUT_HDMI" role="sink">
+ </devicePort>
+ <devicePort tagName="Hdmi-In Mic" type="AUDIO_DEVICE_IN_HDMI" role="source">
+ </devicePort>
+ <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO"
+ role="sink" address="hfp_client_out">
+ </devicePort>
+ <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET"
+ role="source" address="hfp_client_in">
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="Speaker"
+ sources="primary output"/>
+ <route type="mix" sink="primary input"
+ sources="Built-In Mic,Hdmi-In Mic"/>
+ <route type="mix" sink="Hdmi"
+ sources="primary output"/>
+ <route type="mix" sink="BT SCO"
+ sources="mixport_bt_hfp_output"/>
+ <route type="mix" sink="mixport_bt_hfp_input"
+ sources="BT SCO Headset Mic"/>
+ </routes>
+ </module>
+
+ <!-- Remote Submix module -->
+ <module name="r_submix" halVersion="2.0">
+ <attachedDevices>
+ <item>Remote Submix In</item>
+ </attachedDevices>
+ <mixPorts>
+ <mixPort name="r_submix output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="r_submix input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="Remote Submix Out" type="AUDIO_DEVICE_OUT_REMOTE_SUBMIX" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="Remote Submix In" type="AUDIO_DEVICE_IN_REMOTE_SUBMIX" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="Remote Submix Out"
+ sources="r_submix output"/>
+ <route type="mix" sink="r_submix input"
+ sources="Remote Submix In"/>
+ </routes>
+ </module>
+ </modules>
+</audioPolicyConfiguration>
diff --git a/services/audiopolicy/fuzzer/resources/test_audio_policy_msd_configuration.xml b/services/audiopolicy/fuzzer/resources/test_audio_policy_msd_configuration.xml
new file mode 100644
index 0000000..5248d79
--- /dev/null
+++ b/services/audiopolicy/fuzzer/resources/test_audio_policy_msd_configuration.xml
@@ -0,0 +1,84 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2021 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <globalConfiguration speaker_drc_enabled="true"/>
+
+ <modules>
+ <module name="msd" halVersion="2.0">
+ <attachedDevices>
+ <item>MS12 Input</item>
+ <item>MS12 Output</item>
+ </attachedDevices>
+ <mixPorts>
+ <mixPort name="ms12 input" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="ms12 compressed input" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
+ <profile name="" format="AUDIO_FORMAT_AC3"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3_JOC"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_AC4"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_7POINT1"/>
+ </mixPort>
+ <!-- The HW AV Sync flag is not required, but is recommended -->
+ <mixPort name="ms12 output" role="sink" flags="AUDIO_INPUT_FLAG_HW_AV_SYNC|AUDIO_INPUT_FLAG_DIRECT">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ <profile name="" format="AUDIO_FORMAT_AC3"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_5POINT1"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="MS12 Input" type="AUDIO_DEVICE_OUT_BUS" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <profile name="" format="AUDIO_FORMAT_AC3"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3_JOC"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_AC4"
+ samplingRates="32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_5POINT1|AUDIO_CHANNEL_OUT_7POINT1"/>
+ </devicePort>
+ <devicePort tagName="MS12 Output" type="AUDIO_DEVICE_IN_BUS" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="MS12 Input" sources="ms12 input,ms12 compressed input"/>
+ <route type="mix" sink="ms12 output" sources="MS12 Output"/>
+ </routes>
+ </module>
+ </modules>
+</audioPolicyConfiguration>
diff --git a/services/audiopolicy/fuzzer/resources/test_audio_policy_primary_only_configuration.xml b/services/audiopolicy/fuzzer/resources/test_audio_policy_primary_only_configuration.xml
new file mode 100644
index 0000000..15e3773
--- /dev/null
+++ b/services/audiopolicy/fuzzer/resources/test_audio_policy_primary_only_configuration.xml
@@ -0,0 +1,53 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2021 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <globalConfiguration speaker_drc_enabled="true"/>
+
+ <modules>
+ <!-- Primary module -->
+ <module name="primary" halVersion="2.0">
+ <attachedDevices>
+ <item>Speaker</item>
+ <item>Built-In Mic</item>
+ </attachedDevices>
+ <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <mixPorts>
+ <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="primary input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+ </devicePort>
+ <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="Speaker"
+ sources="primary output"/>
+ <route type="mix" sink="primary input"
+ sources="Built-In Mic"/>
+ </routes>
+ </module>
+ </modules>
+</audioPolicyConfiguration>
diff --git a/services/audiopolicy/fuzzer/resources/test_tv_apm_configuration.xml b/services/audiopolicy/fuzzer/resources/test_tv_apm_configuration.xml
new file mode 100644
index 0000000..658d3ce
--- /dev/null
+++ b/services/audiopolicy/fuzzer/resources/test_tv_apm_configuration.xml
@@ -0,0 +1,58 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2021 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <globalConfiguration speaker_drc_enabled="false"/>
+ <modules>
+ <module name="primary" halVersion="2.0">
+ <attachedDevices>
+ <item>Speaker</item>
+ </attachedDevices>
+ <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <mixPorts>
+ <!-- Profiles on the HDMI port are explicit for simplicity. In reality they are dynamic -->
+ <!-- Note: ports are intentionally arranged from more specific to less
+ specific in order to test b/140447125 for HW AV Sync, and similar "explicit matches" -->
+ <mixPort name="tunnel" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_HW_AV_SYNC">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="low latency" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_MMAP_NOIRQ">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="direct" role="source" flags="AUDIO_OUTPUT_FLAG_DIRECT">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink" />
+ <devicePort tagName="Out Aux Digital" type="AUDIO_DEVICE_OUT_AUX_DIGITAL" role="sink" />
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="Speaker" sources="primary output"/>
+ <route type="mix" sink="Out Aux Digital" sources="primary output,tunnel,direct,low latency"/>
+ </routes>
+ </module>
+ </modules>
+</audioPolicyConfiguration>
diff --git a/services/tuner/TunerFrontend.cpp b/services/tuner/TunerFrontend.cpp
index bb8b07d..9b847ab 100644
--- a/services/tuner/TunerFrontend.cpp
+++ b/services/tuner/TunerFrontend.cpp
@@ -20,6 +20,7 @@
using ::aidl::android::media::tv::tuner::TunerFrontendAtsc3PlpSettings;
using ::aidl::android::media::tv::tuner::TunerFrontendScanAtsc3PlpInfo;
+using ::aidl::android::media::tv::tuner::TunerFrontendUnionSettings;
using ::android::hardware::tv::tuner::V1_0::FrontendAnalogSifStandard;
using ::android::hardware::tv::tuner::V1_0::FrontendAnalogType;
using ::android::hardware::tv::tuner::V1_0::FrontendAtscModulation;
@@ -67,7 +68,18 @@
using ::android::hardware::tv::tuner::V1_0::FrontendScanAtsc3PlpInfo;
using ::android::hardware::tv::tuner::V1_0::FrontendScanType;
using ::android::hardware::tv::tuner::V1_0::Result;
+using ::android::hardware::tv::tuner::V1_1::FrontendAnalogAftFlag;
+using ::android::hardware::tv::tuner::V1_1::FrontendCableTimeInterleaveMode;
+using ::android::hardware::tv::tuner::V1_1::FrontendDvbcBandwidth;
using ::android::hardware::tv::tuner::V1_1::FrontendModulation;
+using ::android::hardware::tv::tuner::V1_1::FrontendDtmbBandwidth;
+using ::android::hardware::tv::tuner::V1_1::FrontendDtmbCodeRate;
+using ::android::hardware::tv::tuner::V1_1::FrontendDtmbGuardInterval;
+using ::android::hardware::tv::tuner::V1_1::FrontendDtmbModulation;
+using ::android::hardware::tv::tuner::V1_1::FrontendDtmbTimeInterleaveMode;
+using ::android::hardware::tv::tuner::V1_1::FrontendDtmbTransmissionMode;
+using ::android::hardware::tv::tuner::V1_1::FrontendDvbsScanType;
+using ::android::hardware::tv::tuner::V1_1::FrontendSpectralInversion;
namespace android {
@@ -105,12 +117,23 @@
Status TunerFrontend::tune(const TunerFrontendSettings& settings) {
if (mFrontend == NULL) {
- ALOGD("IFrontend is not initialized");
+ ALOGE("IFrontend is not initialized");
return Status::fromServiceSpecificError(static_cast<int32_t>(Result::UNAVAILABLE));
}
+ Result status;
FrontendSettings frontendSettings = getHidlFrontendSettings(settings);
- Result status = mFrontend->tune(frontendSettings);
+ if (settings.isExtended) {
+ if (mFrontend_1_1 == NULL) {
+ ALOGE("IFrontend_1_1 is not initialized");
+ return Status::fromServiceSpecificError(static_cast<int32_t>(Result::UNAVAILABLE));
+ }
+ FrontendSettingsExt1_1 frontendSettingsExt = getHidlFrontendSettingsExt(settings);
+ status = mFrontend_1_1->tune_1_1(frontendSettings, frontendSettingsExt);
+ } else {
+ status = mFrontend->tune(frontendSettings);
+ }
+
if (status == Result::SUCCESS) {
return Status::ok();
}
@@ -138,9 +161,21 @@
return Status::fromServiceSpecificError(static_cast<int32_t>(Result::UNAVAILABLE));
}
+ Result status;
FrontendSettings frontendSettings = getHidlFrontendSettings(settings);
- Result status = mFrontend->scan(
- frontendSettings, static_cast<FrontendScanType>(frontendScanType));
+ if (settings.isExtended) {
+ if (mFrontend_1_1 == NULL) {
+ ALOGE("IFrontend_1_1 is not initialized");
+ return Status::fromServiceSpecificError(static_cast<int32_t>(Result::UNAVAILABLE));
+ }
+ FrontendSettingsExt1_1 frontendSettingsExt = getHidlFrontendSettingsExt(settings);
+ status = mFrontend_1_1->scan_1_1(frontendSettings,
+ static_cast<FrontendScanType>(frontendScanType), frontendSettingsExt);
+ } else {
+ status = mFrontend->scan(
+ frontendSettings, static_cast<FrontendScanType>(frontendScanType));
+ }
+
if (status == Result::SUCCESS) {
return Status::ok();
}
@@ -303,24 +338,34 @@
case FrontendScanMessageTypeExt1_1::MODULATION: {
FrontendModulation m = message.modulation();
int modulation;
- if (m.getDiscriminator() == FrontendModulation::hidl_discriminator::dvbc) {
- modulation = (int) m.dvbc();
- } else if (m.getDiscriminator() == FrontendModulation::hidl_discriminator::dvbt) {
- modulation = (int) m.dvbt();
- } else if (m.getDiscriminator() == FrontendModulation::hidl_discriminator::dvbs) {
- modulation = (int) m.dvbs();
- } else if (m.getDiscriminator() == FrontendModulation::hidl_discriminator::isdbs) {
- modulation = (int) m.isdbs();
- } else if (m.getDiscriminator() == FrontendModulation::hidl_discriminator::isdbs3) {
- modulation = (int) m.isdbs3();
- } else if (m.getDiscriminator() == FrontendModulation::hidl_discriminator::isdbt) {
- modulation = (int) m.isdbt();
- } else if (m.getDiscriminator() == FrontendModulation::hidl_discriminator::atsc) {
- modulation = (int) m.atsc();
- } else if (m.getDiscriminator() == FrontendModulation::hidl_discriminator::atsc3) {
- modulation = (int) m.atsc3();
- } else if (m.getDiscriminator() == FrontendModulation::hidl_discriminator::dtmb) {
- modulation = (int) m.dtmb();
+ switch (m.getDiscriminator()) {
+ case FrontendModulation::hidl_discriminator::dvbc:
+ modulation = (int) m.dvbc();
+ break;
+ case FrontendModulation::hidl_discriminator::dvbt:
+ modulation = (int) m.dvbt();
+ break;
+ case FrontendModulation::hidl_discriminator::dvbs:
+ modulation = (int) m.dvbs();
+ break;
+ case FrontendModulation::hidl_discriminator::isdbs:
+ modulation = (int) m.isdbs();
+ break;
+ case FrontendModulation::hidl_discriminator::isdbs3:
+ modulation = (int) m.isdbs3();
+ break;
+ case FrontendModulation::hidl_discriminator::isdbt:
+ modulation = (int) m.isdbt();
+ break;
+ case FrontendModulation::hidl_discriminator::atsc:
+ modulation = (int) m.atsc();
+ break;
+ case FrontendModulation::hidl_discriminator::atsc3:
+ modulation = (int) m.atsc3();
+ break;
+ case FrontendModulation::hidl_discriminator::dtmb:
+ modulation = (int) m.dtmb();
+ break;
}
scanMessage.set<TunerFrontendScanMessage::modulation>(modulation);
break;
@@ -384,166 +429,216 @@
return coderate;
}
-FrontendSettings TunerFrontend::getHidlFrontendSettings(const TunerFrontendSettings& settings) {
- // TODO: extend TunerFrontendSettings to use 1.1 types
+FrontendSettings TunerFrontend::getHidlFrontendSettings(const TunerFrontendSettings& aidlSettings) {
+ auto settings = aidlSettings.settings;
FrontendSettings frontendSettings;
+
switch (settings.getTag()) {
- case TunerFrontendSettings::analog:
+ case TunerFrontendUnionSettings::analog: {
+ auto analog = settings.get<TunerFrontendUnionSettings::analog>();
frontendSettings.analog({
- .frequency = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::analog>().frequency),
- .type = static_cast<FrontendAnalogType>(
- settings.get<TunerFrontendSettings::analog>().signalType),
- .sifStandard = static_cast<FrontendAnalogSifStandard>(
- settings.get<TunerFrontendSettings::analog>().sifStandard),
+ .frequency = static_cast<uint32_t>(analog.frequency),
+ .type = static_cast<FrontendAnalogType>(analog.signalType),
+ .sifStandard = static_cast<FrontendAnalogSifStandard>(analog.sifStandard),
});
break;
- case TunerFrontendSettings::atsc:
+ }
+ case TunerFrontendUnionSettings::atsc: {
+ auto atsc = settings.get<TunerFrontendUnionSettings::atsc>();
frontendSettings.atsc({
- .frequency = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::atsc>().frequency),
- .modulation = static_cast<FrontendAtscModulation>(
- settings.get<TunerFrontendSettings::atsc>().modulation),
+ .frequency = static_cast<uint32_t>(atsc.frequency),
+ .modulation = static_cast<FrontendAtscModulation>(atsc.modulation),
});
break;
- case TunerFrontendSettings::atsc3:
+ }
+ case TunerFrontendUnionSettings::atsc3: {
+ auto atsc3 = settings.get<TunerFrontendUnionSettings::atsc3>();
frontendSettings.atsc3({
- .frequency = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::atsc3>().frequency),
- .bandwidth = static_cast<FrontendAtsc3Bandwidth>(
- settings.get<TunerFrontendSettings::atsc3>().bandwidth),
+ .frequency = static_cast<uint32_t>(atsc3.frequency),
+ .bandwidth = static_cast<FrontendAtsc3Bandwidth>(atsc3.bandwidth),
.demodOutputFormat = static_cast<FrontendAtsc3DemodOutputFormat>(
- settings.get<TunerFrontendSettings::atsc3>().demodOutputFormat),
- .plpSettings = getAtsc3PlpSettings(settings.get<TunerFrontendSettings::atsc3>()),
+ atsc3.demodOutputFormat),
+ .plpSettings = getAtsc3PlpSettings(atsc3),
});
break;
- case TunerFrontendSettings::cable:
+ }
+ case TunerFrontendUnionSettings::cable: {
+ auto dvbc = settings.get<TunerFrontendUnionSettings::cable>();
frontendSettings.dvbc({
- .frequency = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::cable>().frequency),
- .modulation = static_cast<FrontendDvbcModulation>(
- settings.get<TunerFrontendSettings::cable>().modulation),
- .fec = static_cast<FrontendInnerFec>(
- settings.get<TunerFrontendSettings::cable>().innerFec),
- .symbolRate = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::cable>().symbolRate),
- .outerFec = static_cast<FrontendDvbcOuterFec>(
- settings.get<TunerFrontendSettings::cable>().outerFec),
- .annex = static_cast<FrontendDvbcAnnex>(
- settings.get<TunerFrontendSettings::cable>().annex),
+ .frequency = static_cast<uint32_t>(dvbc.frequency),
+ .modulation = static_cast<FrontendDvbcModulation>(dvbc.modulation),
+ .fec = static_cast<FrontendInnerFec>(dvbc.innerFec),
+ .symbolRate = static_cast<uint32_t>(dvbc.symbolRate),
+ .outerFec = static_cast<FrontendDvbcOuterFec>(dvbc.outerFec),
+ .annex = static_cast<FrontendDvbcAnnex>(dvbc.annex),
.spectralInversion = static_cast<FrontendDvbcSpectralInversion>(
- settings.get<TunerFrontendSettings::cable>().spectralInversion),
+ dvbc.spectralInversion),
});
break;
- case TunerFrontendSettings::dvbs:
+ }
+ case TunerFrontendUnionSettings::dvbs: {
+ auto dvbs = settings.get<TunerFrontendUnionSettings::dvbs>();
frontendSettings.dvbs({
- .frequency = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::dvbs>().frequency),
- .modulation = static_cast<FrontendDvbsModulation>(
- settings.get<TunerFrontendSettings::dvbs>().modulation),
- .coderate = getDvbsCodeRate(
- settings.get<TunerFrontendSettings::dvbs>().codeRate),
- .symbolRate = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::dvbs>().symbolRate),
- .rolloff = static_cast<FrontendDvbsRolloff>(
- settings.get<TunerFrontendSettings::dvbs>().rolloff),
- .pilot = static_cast<FrontendDvbsPilot>(
- settings.get<TunerFrontendSettings::dvbs>().pilot),
- .inputStreamId = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::dvbs>().inputStreamId),
- .standard = static_cast<FrontendDvbsStandard>(
- settings.get<TunerFrontendSettings::dvbs>().standard),
- .vcmMode = static_cast<FrontendDvbsVcmMode>(
- settings.get<TunerFrontendSettings::dvbs>().vcm),
+ .frequency = static_cast<uint32_t>(dvbs.frequency),
+ .modulation = static_cast<FrontendDvbsModulation>(dvbs.modulation),
+ .coderate = getDvbsCodeRate(dvbs.codeRate),
+ .symbolRate = static_cast<uint32_t>(dvbs.symbolRate),
+ .rolloff = static_cast<FrontendDvbsRolloff>(dvbs.rolloff),
+ .pilot = static_cast<FrontendDvbsPilot>(dvbs.pilot),
+ .inputStreamId = static_cast<uint32_t>(dvbs.inputStreamId),
+ .standard = static_cast<FrontendDvbsStandard>(dvbs.standard),
+ .vcmMode = static_cast<FrontendDvbsVcmMode>(dvbs.vcm),
});
break;
- case TunerFrontendSettings::dvbt:
+ }
+ case TunerFrontendUnionSettings::dvbt: {
+ auto dvbt = settings.get<TunerFrontendUnionSettings::dvbt>();
frontendSettings.dvbt({
- .frequency = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::dvbt>().frequency),
+ .frequency = static_cast<uint32_t>(dvbt.frequency),
.transmissionMode = static_cast<FrontendDvbtTransmissionMode>(
- settings.get<TunerFrontendSettings::dvbt>().transmissionMode),
- .bandwidth = static_cast<FrontendDvbtBandwidth>(
- settings.get<TunerFrontendSettings::dvbt>().bandwidth),
- .constellation = static_cast<FrontendDvbtConstellation>(
- settings.get<TunerFrontendSettings::dvbt>().constellation),
- .hierarchy = static_cast<FrontendDvbtHierarchy>(
- settings.get<TunerFrontendSettings::dvbt>().hierarchy),
- .hpCoderate = static_cast<FrontendDvbtCoderate>(
- settings.get<TunerFrontendSettings::dvbt>().hpCodeRate),
- .lpCoderate = static_cast<FrontendDvbtCoderate>(
- settings.get<TunerFrontendSettings::dvbt>().lpCodeRate),
- .guardInterval = static_cast<FrontendDvbtGuardInterval>(
- settings.get<TunerFrontendSettings::dvbt>().guardInterval),
- .isHighPriority = settings.get<TunerFrontendSettings::dvbt>().isHighPriority,
- .standard = static_cast<FrontendDvbtStandard>(
- settings.get<TunerFrontendSettings::dvbt>().standard),
- .isMiso = settings.get<TunerFrontendSettings::dvbt>().isMiso,
- .plpMode = static_cast<FrontendDvbtPlpMode>(
- settings.get<TunerFrontendSettings::dvbt>().plpMode),
- .plpId = static_cast<uint8_t>(
- settings.get<TunerFrontendSettings::dvbt>().plpId),
- .plpGroupId = static_cast<uint8_t>(
- settings.get<TunerFrontendSettings::dvbt>().plpGroupId),
+ dvbt.transmissionMode),
+ .bandwidth = static_cast<FrontendDvbtBandwidth>(dvbt.bandwidth),
+ .constellation = static_cast<FrontendDvbtConstellation>(dvbt.constellation),
+ .hierarchy = static_cast<FrontendDvbtHierarchy>(dvbt.hierarchy),
+ .hpCoderate = static_cast<FrontendDvbtCoderate>(dvbt.hpCodeRate),
+ .lpCoderate = static_cast<FrontendDvbtCoderate>(dvbt.lpCodeRate),
+ .guardInterval = static_cast<FrontendDvbtGuardInterval>(dvbt.guardInterval),
+ .isHighPriority = dvbt.isHighPriority,
+ .standard = static_cast<FrontendDvbtStandard>(dvbt.standard),
+ .isMiso = dvbt.isMiso,
+ .plpMode = static_cast<FrontendDvbtPlpMode>(dvbt.plpMode),
+ .plpId = static_cast<uint8_t>(dvbt.plpId),
+ .plpGroupId = static_cast<uint8_t>(dvbt.plpGroupId),
});
break;
- case TunerFrontendSettings::isdbs:
+ }
+ case TunerFrontendUnionSettings::isdbs: {
+ auto isdbs = settings.get<TunerFrontendUnionSettings::isdbs>();
frontendSettings.isdbs({
- .frequency = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::isdbs>().frequency),
- .streamId = static_cast<uint16_t>(
- settings.get<TunerFrontendSettings::isdbs>().streamId),
- .streamIdType = static_cast<FrontendIsdbsStreamIdType>(
- settings.get<TunerFrontendSettings::isdbs>().streamIdType),
- .modulation = static_cast<FrontendIsdbsModulation>(
- settings.get<TunerFrontendSettings::isdbs>().modulation),
- .coderate = static_cast<FrontendIsdbsCoderate>(
- settings.get<TunerFrontendSettings::isdbs>().codeRate),
- .symbolRate = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::isdbs>().symbolRate),
- .rolloff = static_cast<FrontendIsdbsRolloff>(
- settings.get<TunerFrontendSettings::isdbs>().rolloff),
+ .frequency = static_cast<uint32_t>(isdbs.frequency),
+ .streamId = static_cast<uint16_t>(isdbs.streamId),
+ .streamIdType = static_cast<FrontendIsdbsStreamIdType>(isdbs.streamIdType),
+ .modulation = static_cast<FrontendIsdbsModulation>(isdbs.modulation),
+ .coderate = static_cast<FrontendIsdbsCoderate>(isdbs.codeRate),
+ .symbolRate = static_cast<uint32_t>(isdbs.symbolRate),
+ .rolloff = static_cast<FrontendIsdbsRolloff>(isdbs.rolloff),
});
break;
- case TunerFrontendSettings::isdbs3:
+ }
+ case TunerFrontendUnionSettings::isdbs3: {
+ auto isdbs3 = settings.get<TunerFrontendUnionSettings::isdbs3>();
frontendSettings.isdbs3({
- .frequency = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::isdbs3>().frequency),
- .streamId = static_cast<uint16_t>(
- settings.get<TunerFrontendSettings::isdbs3>().streamId),
- .streamIdType = static_cast<FrontendIsdbsStreamIdType>(
- settings.get<TunerFrontendSettings::isdbs3>().streamIdType),
- .modulation = static_cast<FrontendIsdbs3Modulation>(
- settings.get<TunerFrontendSettings::isdbs3>().modulation),
- .coderate = static_cast<FrontendIsdbs3Coderate>(
- settings.get<TunerFrontendSettings::isdbs3>().codeRate),
- .symbolRate = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::isdbs3>().symbolRate),
- .rolloff = static_cast<FrontendIsdbs3Rolloff>(
- settings.get<TunerFrontendSettings::isdbs3>().rolloff),
+ .frequency = static_cast<uint32_t>(isdbs3.frequency),
+ .streamId = static_cast<uint16_t>(isdbs3.streamId),
+ .streamIdType = static_cast<FrontendIsdbsStreamIdType>(isdbs3.streamIdType),
+ .modulation = static_cast<FrontendIsdbs3Modulation>(isdbs3.modulation),
+ .coderate = static_cast<FrontendIsdbs3Coderate>(isdbs3.codeRate),
+ .symbolRate = static_cast<uint32_t>(isdbs3.symbolRate),
+ .rolloff = static_cast<FrontendIsdbs3Rolloff>(isdbs3.rolloff),
});
break;
- case TunerFrontendSettings::isdbt:
+ }
+ case TunerFrontendUnionSettings::isdbt: {
+ auto isdbt = settings.get<TunerFrontendUnionSettings::isdbt>();
frontendSettings.isdbt({
- .frequency = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::isdbt>().frequency),
- .modulation = static_cast<FrontendIsdbtModulation>(
- settings.get<TunerFrontendSettings::isdbt>().modulation),
- .bandwidth = static_cast<FrontendIsdbtBandwidth>(
- settings.get<TunerFrontendSettings::isdbt>().bandwidth),
- .mode = static_cast<FrontendIsdbtMode>(
- settings.get<TunerFrontendSettings::isdbt>().mode),
- .coderate = static_cast<FrontendIsdbtCoderate>(
- settings.get<TunerFrontendSettings::isdbt>().codeRate),
- .guardInterval = static_cast<FrontendIsdbtGuardInterval>(
- settings.get<TunerFrontendSettings::isdbt>().guardInterval),
- .serviceAreaId = static_cast<uint32_t>(
- settings.get<TunerFrontendSettings::isdbt>().serviceAreaId),
+ .frequency = static_cast<uint32_t>(isdbt.frequency),
+ .modulation = static_cast<FrontendIsdbtModulation>(isdbt.modulation),
+ .bandwidth = static_cast<FrontendIsdbtBandwidth>(isdbt.bandwidth),
+ .mode = static_cast<FrontendIsdbtMode>(isdbt.mode),
+ .coderate = static_cast<FrontendIsdbtCoderate>(isdbt.codeRate),
+ .guardInterval = static_cast<FrontendIsdbtGuardInterval>(isdbt.guardInterval),
+ .serviceAreaId = static_cast<uint32_t>(isdbt.serviceAreaId),
});
break;
+ }
default:
break;
}
+
return frontendSettings;
}
+
+FrontendSettingsExt1_1 TunerFrontend::getHidlFrontendSettingsExt(
+ const TunerFrontendSettings& aidlSettings) {
+ FrontendSettingsExt1_1 frontendSettingsExt{
+ .endFrequency = static_cast<uint32_t>(aidlSettings.endFrequency),
+ .inversion = static_cast<FrontendSpectralInversion>(aidlSettings.inversion),
+ };
+
+ auto settings = aidlSettings.settings;
+ switch (settings.getTag()) {
+ case TunerFrontendUnionSettings::analog: {
+ auto analog = settings.get<TunerFrontendUnionSettings::analog>();
+ if (analog.isExtended) {
+ frontendSettingsExt.settingExt.analog({
+ .aftFlag = static_cast<FrontendAnalogAftFlag>(analog.aftFlag),
+ });
+ } else {
+ frontendSettingsExt.settingExt.noinit();
+ }
+ break;
+ }
+ case TunerFrontendUnionSettings::cable: {
+ auto dvbc = settings.get<TunerFrontendUnionSettings::cable>();
+ if (dvbc.isExtended) {
+ frontendSettingsExt.settingExt.dvbc({
+ .interleaveMode = static_cast<FrontendCableTimeInterleaveMode>(
+ dvbc.interleaveMode),
+ .bandwidth = static_cast<FrontendDvbcBandwidth>(
+ dvbc.bandwidth),
+ });
+ } else {
+ frontendSettingsExt.settingExt.noinit();
+ }
+ break;
+ }
+ case TunerFrontendUnionSettings::dvbs: {
+ auto dvbs = settings.get<TunerFrontendUnionSettings::dvbs>();
+ if (dvbs.isExtended) {
+ frontendSettingsExt.settingExt.dvbs({
+ .scanType = static_cast<FrontendDvbsScanType>(dvbs.scanType),
+ .isDiseqcRxMessage = dvbs.isDiseqcRxMessage,
+ });
+ } else {
+ frontendSettingsExt.settingExt.noinit();
+ }
+ break;
+ }
+ case TunerFrontendUnionSettings::dvbt: {
+ auto dvbt = settings.get<TunerFrontendUnionSettings::dvbt>();
+ if (dvbt.isExtended) {
+ frontendSettingsExt.settingExt.dvbt({
+ .constellation =
+ static_cast<hardware::tv::tuner::V1_1::FrontendDvbtConstellation>(
+ dvbt.constellation),
+ .transmissionMode =
+ static_cast<hardware::tv::tuner::V1_1::FrontendDvbtTransmissionMode>(
+ dvbt.transmissionMode),
+ });
+ } else {
+ frontendSettingsExt.settingExt.noinit();
+ }
+ break;
+ }
+ case TunerFrontendUnionSettings::dtmb: {
+ auto dtmb = settings.get<TunerFrontendUnionSettings::dtmb>();
+ frontendSettingsExt.settingExt.dtmb({
+ .frequency = static_cast<uint32_t>(dtmb.frequency),
+ .transmissionMode = static_cast<FrontendDtmbTransmissionMode>(
+ dtmb.transmissionMode),
+ .bandwidth = static_cast<FrontendDtmbBandwidth>(dtmb.bandwidth),
+ .modulation = static_cast<FrontendDtmbModulation>(dtmb.modulation),
+ .codeRate = static_cast<FrontendDtmbCodeRate>(dtmb.codeRate),
+ .guardInterval = static_cast<FrontendDtmbGuardInterval>(dtmb.guardInterval),
+ .interleaveMode = static_cast<FrontendDtmbTimeInterleaveMode>(dtmb.interleaveMode),
+ });
+ break;
+ }
+ default:
+ frontendSettingsExt.settingExt.noinit();
+ break;
+ }
+
+ return frontendSettingsExt;
+}
} // namespace android
diff --git a/services/tuner/TunerFrontend.h b/services/tuner/TunerFrontend.h
index 431022d..c58a862 100644
--- a/services/tuner/TunerFrontend.h
+++ b/services/tuner/TunerFrontend.h
@@ -41,11 +41,12 @@
using ::android::hardware::tv::tuner::V1_0::FrontendId;
using ::android::hardware::tv::tuner::V1_0::FrontendScanMessage;
using ::android::hardware::tv::tuner::V1_0::FrontendScanMessageType;
-using ::android::hardware::tv::tuner::V1_0::FrontendSettings;;
+using ::android::hardware::tv::tuner::V1_0::FrontendSettings;
using ::android::hardware::tv::tuner::V1_0::IFrontend;
using ::android::hardware::tv::tuner::V1_1::IFrontendCallback;
using ::android::hardware::tv::tuner::V1_1::FrontendScanMessageExt1_1;
using ::android::hardware::tv::tuner::V1_1::FrontendScanMessageTypeExt1_1;
+using ::android::hardware::tv::tuner::V1_1::FrontendSettingsExt1_1;
using namespace std;
@@ -86,7 +87,8 @@
hidl_vec<FrontendAtsc3PlpSettings> getAtsc3PlpSettings(
const TunerFrontendAtsc3Settings& settings);
FrontendDvbsCodeRate getDvbsCodeRate(const TunerFrontendDvbsCodeRate& codeRate);
- FrontendSettings getHidlFrontendSettings(const TunerFrontendSettings& settings);
+ FrontendSettings getHidlFrontendSettings(const TunerFrontendSettings& aidlSettings);
+ FrontendSettingsExt1_1 getHidlFrontendSettingsExt(const TunerFrontendSettings& aidlSettings);
int mId;
sp<IFrontend> mFrontend;
diff --git a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendAnalogSettings.aidl b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendAnalogSettings.aidl
index b6d07c3..40cd8c9 100644
--- a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendAnalogSettings.aidl
+++ b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendAnalogSettings.aidl
@@ -33,4 +33,11 @@
* Standard Interchange Format (SIF) setting
*/
int sifStandard;
+
+ /**
+ * Fields after isExtended are only valid when isExtended is true
+ */
+ boolean isExtended;
+
+ int aftFlag;
}
diff --git a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendCableSettings.aidl b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendCableSettings.aidl
index 3984f2c..b9bcf29 100644
--- a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendCableSettings.aidl
+++ b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendCableSettings.aidl
@@ -51,4 +51,13 @@
* Spectral Inversion Type.
*/
int spectralInversion;
+
+ /**
+ * Fields after isExtended are only valid when isExtended is true
+ */
+ boolean isExtended;
+
+ int interleaveMode;
+
+ int bandwidth;
}
diff --git a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendDtmbSettings.aidl b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendDtmbSettings.aidl
new file mode 100644
index 0000000..45e7ff9
--- /dev/null
+++ b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendDtmbSettings.aidl
@@ -0,0 +1,38 @@
+/**
+ * Copyright 2021, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media.tv.tuner;
+
+/**
+ * DTMB Frontend Settings interface.
+ *
+ * {@hide}
+ */
+parcelable TunerFrontendDtmbSettings {
+ int frequency;
+
+ int transmissionMode;
+
+ int bandwidth;
+
+ int modulation;
+
+ int codeRate;
+
+ int guardInterval;
+
+ int interleaveMode;
+}
diff --git a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendDvbsSettings.aidl b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendDvbsSettings.aidl
index 554a502..ec3e4b9 100644
--- a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendDvbsSettings.aidl
+++ b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendDvbsSettings.aidl
@@ -53,4 +53,13 @@
* Vcm mode.
*/
int vcm;
+
+ /**
+ * Fields after isExtended are only valid when isExtended is true
+ */
+ boolean isExtended;
+
+ int scanType;
+
+ boolean isDiseqcRxMessage;
}
diff --git a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendDvbtSettings.aidl b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendDvbtSettings.aidl
index c72396b..14c942a 100644
--- a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendDvbtSettings.aidl
+++ b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendDvbtSettings.aidl
@@ -67,4 +67,9 @@
* Physical Layer Pipe (PLP) Group Id
*/
int plpGroupId;
+
+ /**
+ * Fields after isExtended are only valid when isExtended is true
+ */
+ boolean isExtended;
}
diff --git a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendSettings.aidl b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendSettings.aidl
index a382941..70a5f3e 100644
--- a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendSettings.aidl
+++ b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendSettings.aidl
@@ -1,5 +1,5 @@
/**
- * Copyright 2020, The Android Open Source Project
+ * Copyright 2021, The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -16,37 +16,19 @@
package android.media.tv.tuner;
-import android.media.tv.tuner.TunerFrontendAnalogSettings;
-import android.media.tv.tuner.TunerFrontendAtscSettings;
-import android.media.tv.tuner.TunerFrontendAtsc3Settings;
-import android.media.tv.tuner.TunerFrontendCableSettings;
-import android.media.tv.tuner.TunerFrontendDvbsSettings;
-import android.media.tv.tuner.TunerFrontendDvbtSettings;
-import android.media.tv.tuner.TunerFrontendIsdbsSettings;
-import android.media.tv.tuner.TunerFrontendIsdbs3Settings;
-import android.media.tv.tuner.TunerFrontendIsdbtSettings;
+import android.media.tv.tuner.TunerFrontendUnionSettings;
/**
- * Analog Frontend Settings interface.
+ * Frontend Settings interface.
*
* {@hide}
*/
-union TunerFrontendSettings {
- TunerFrontendAnalogSettings analog;
+parcelable TunerFrontendSettings {
+ TunerFrontendUnionSettings settings;
- TunerFrontendAtscSettings atsc;
+ boolean isExtended;
- TunerFrontendAtsc3Settings atsc3;
+ int endFrequency;
- TunerFrontendCableSettings cable;
-
- TunerFrontendDvbsSettings dvbs;
-
- TunerFrontendDvbtSettings dvbt;
-
- TunerFrontendIsdbsSettings isdbs;
-
- TunerFrontendIsdbs3Settings isdbs3;
-
- TunerFrontendIsdbtSettings isdbt;
+ int inversion;
}
diff --git a/services/tuner/aidl/android/media/tv/tuner/TunerFrontendUnionSettings.aidl b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendUnionSettings.aidl
new file mode 100644
index 0000000..c362c2a
--- /dev/null
+++ b/services/tuner/aidl/android/media/tv/tuner/TunerFrontendUnionSettings.aidl
@@ -0,0 +1,55 @@
+/**
+ * Copyright 2020, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media.tv.tuner;
+
+import android.media.tv.tuner.TunerFrontendAnalogSettings;
+import android.media.tv.tuner.TunerFrontendAtscSettings;
+import android.media.tv.tuner.TunerFrontendAtsc3Settings;
+import android.media.tv.tuner.TunerFrontendCableSettings;
+import android.media.tv.tuner.TunerFrontendDtmbSettings;
+import android.media.tv.tuner.TunerFrontendDvbsSettings;
+import android.media.tv.tuner.TunerFrontendDvbtSettings;
+import android.media.tv.tuner.TunerFrontendIsdbsSettings;
+import android.media.tv.tuner.TunerFrontendIsdbs3Settings;
+import android.media.tv.tuner.TunerFrontendIsdbtSettings;
+
+/**
+ * Frontend Settings Union interface.
+ *
+ * {@hide}
+ */
+union TunerFrontendUnionSettings {
+ TunerFrontendAnalogSettings analog;
+
+ TunerFrontendAtscSettings atsc;
+
+ TunerFrontendAtsc3Settings atsc3;
+
+ TunerFrontendCableSettings cable;
+
+ TunerFrontendDvbsSettings dvbs;
+
+ TunerFrontendDvbtSettings dvbt;
+
+ TunerFrontendIsdbsSettings isdbs;
+
+ TunerFrontendIsdbs3Settings isdbs3;
+
+ TunerFrontendIsdbtSettings isdbt;
+
+ TunerFrontendDtmbSettings dtmb;
+}