Merge "AudioFlinger: fallback to non-primary HW devs if getInputBufferSize failed" into main
diff --git a/Android.bp b/Android.bp
index 72b8721..afb1341 100644
--- a/Android.bp
+++ b/Android.bp
@@ -133,3 +133,19 @@
frozen: true,
}
+
+latest_av_audio_types_aidl = "av-audio-types-aidl-V1"
+
+cc_defaults {
+ name: "latest_av_audio_types_aidl_ndk_shared",
+ shared_libs: [
+ latest_av_audio_types_aidl + "-ndk",
+ ],
+}
+
+cc_defaults {
+ name: "latest_av_audio_types_aidl_ndk_static",
+ static_libs: [
+ latest_av_audio_types_aidl + "-ndk",
+ ],
+}
diff --git a/media/TEST_MAPPING b/media/TEST_MAPPING
index d388596..1a637ac 100644
--- a/media/TEST_MAPPING
+++ b/media/TEST_MAPPING
@@ -47,8 +47,13 @@
],
// Postsubmit tests for TV devices
"tv-postsubmit": [
- {
- "name": "DecoderRenderTest"
- }
+ {
+ "name": "CtsMediaDecoderTestCases",
+ "options": [
+ {
+ "include-filter": "android.media.decoder.cts.DecoderRenderTest"
+ }
+ ]
+ }
]
}
diff --git a/media/codec2/hal/client/GraphicsTracker.cpp b/media/codec2/hal/client/GraphicsTracker.cpp
index 1c2a0fb..dbbabfe 100644
--- a/media/codec2/hal/client/GraphicsTracker.cpp
+++ b/media/codec2/hal/client/GraphicsTracker.cpp
@@ -173,7 +173,7 @@
}
GraphicsTracker::GraphicsTracker(int maxDequeueCount)
- : mBufferCache(new BufferCache()), mMaxDequeue{maxDequeueCount},
+ : mBufferCache(new BufferCache()), mNumDequeueing{0}, mMaxDequeue{maxDequeueCount},
mMaxDequeueCommitted{maxDequeueCount},
mDequeueable{maxDequeueCount},
mTotalDequeued{0}, mTotalCancelled{0}, mTotalDropped{0}, mTotalReleased{0},
@@ -235,6 +235,7 @@
const sp<IGraphicBufferProducer>& igbp, uint32_t generation) {
// TODO: wait until operations to previous IGBP is completed.
std::shared_ptr<BufferCache> prevCache;
+ int prevDequeueRequested = 0;
int prevDequeueCommitted;
std::unique_lock<std::mutex> cl(mConfigLock);
@@ -243,6 +244,9 @@
mInConfig = true;
prevCache = mBufferCache;
prevDequeueCommitted = mMaxDequeueCommitted;
+ if (mMaxDequeueRequested.has_value()) {
+ prevDequeueRequested = mMaxDequeueRequested.value();
+ }
}
// NOTE: Switching to the same surface is blocked from MediaCodec.
// Switching to the same surface might not work if tried, since disconnect()
@@ -263,6 +267,11 @@
mInConfig = false;
return C2_BAD_VALUE;
}
+ ALOGD("new surface in configuration: maxDequeueRequested(%d), maxDequeueCommitted(%d)",
+ prevDequeueRequested, prevDequeueCommitted);
+ if (prevDequeueRequested > 0 && prevDequeueRequested > prevDequeueCommitted) {
+ prevDequeueCommitted = prevDequeueRequested;
+ }
if (igbp) {
ret = igbp->setMaxDequeuedBufferCount(prevDequeueCommitted);
if (ret != ::android::OK) {
@@ -280,6 +289,34 @@
std::unique_lock<std::mutex> l(mLock);
mInConfig = false;
mBufferCache = newCache;
+ // {@code dequeued} is the number of currently dequeued buffers.
+ // {@code prevDequeueCommitted} is max dequeued buffer at any moment
+ // from the new surface.
+ // {@code newDequeueable} is hence the current # of dequeueable buffers
+ // if no change occurs.
+ int dequeued = mDequeued.size() + mNumDequeueing;
+ int newDequeueable = prevDequeueCommitted - dequeued;
+ if (newDequeueable < 0) {
+ // This will not happen.
+ // But if this happens, we respect the value and try to continue.
+ ALOGE("calculated new dequeueable is negative: %d max(%d),dequeued(%d)",
+ newDequeueable, prevDequeueCommitted, dequeued);
+ }
+
+ if (mMaxDequeueRequested.has_value() && mMaxDequeueRequested == prevDequeueCommitted) {
+ mMaxDequeueRequested.reset();
+ }
+ mMaxDequeue = mMaxDequeueCommitted = prevDequeueCommitted;
+
+ int delta = newDequeueable - mDequeueable;
+ if (delta > 0) {
+ writeIncDequeueableLocked(delta);
+ } else if (delta < 0) {
+ drainDequeueableLocked(-delta);
+ }
+ ALOGV("new surfcace dequeueable %d(delta %d), maxDequeue %d",
+ newDequeueable, delta, mMaxDequeue);
+ mDequeueable = newDequeueable;
}
return C2_OK;
}
@@ -529,6 +566,7 @@
ALOGE("writing end for the waitable object seems to be closed");
return C2_BAD_STATE;
}
+ mNumDequeueing++;
mDequeueable--;
*cache = mBufferCache;
return C2_OK;
@@ -543,6 +581,7 @@
bool cached, int slot, const sp<Fence> &fence,
std::shared_ptr<BufferItem> *pBuffer, bool *updateDequeue) {
std::unique_lock<std::mutex> l(mLock);
+ mNumDequeueing--;
if (res == C2_OK) {
if (cached) {
auto it = cache->mBuffers.find(slot);
@@ -655,7 +694,8 @@
ALOGE("allocate by dequeueBuffer() successful, but requestBuffer() failed %d",
status);
igbp->cancelBuffer(slotId, fence);
- return C2_CORRUPTED;
+ // This might be due to life-cycle end and/or surface switching.
+ return C2_BLOCKING;
}
*buffer = std::make_shared<BufferItem>(generation, slotId, realloced, fence);
if (!*buffer) {
diff --git a/media/codec2/hal/client/include/codec2/aidl/GraphicsTracker.h b/media/codec2/hal/client/include/codec2/aidl/GraphicsTracker.h
index dd6c869..762030b 100644
--- a/media/codec2/hal/client/include/codec2/aidl/GraphicsTracker.h
+++ b/media/codec2/hal/client/include/codec2/aidl/GraphicsTracker.h
@@ -234,6 +234,7 @@
// Maps bufferId to buffer
std::map<uint64_t, std::shared_ptr<BufferItem>> mDequeued;
std::set<uint64_t> mDeallocating;
+ int mNumDequeueing;
// These member variables are read and modified accessed as follows.
// 1. mConfigLock being held
diff --git a/media/codec2/sfplugin/CCodec.cpp b/media/codec2/sfplugin/CCodec.cpp
index 20b6d7f..a897fa0 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -2229,9 +2229,15 @@
// See also b/300350761.
//
// The workaround is no longer needed with fetchGraphicBlock & C2Fence changes.
- // so we are reverting back to the logical sequence of the operations.
+ // so we are reverting back to the logical sequence of the operations when
+ // AIDL HALs are selected.
+ // When the HIDL HALs are selected, we retained workaround(the reversed
+ // order) as default in order to keep legacy behavior.
+ bool stopHalBeforeSurface =
+ Codec2Client::IsAidlSelected() ||
+ property_get_bool("debug.codec2.stop_hal_before_surface", false);
status_t err = C2_OK;
- if (android::media::codec::provider_->stop_hal_before_surface()) {
+ if (stopHalBeforeSurface && android::media::codec::provider_->stop_hal_before_surface()) {
err = comp->stop();
mChannel->stopUseOutputSurface(pushBlankBuffer);
} else {
@@ -2334,8 +2340,14 @@
// See also b/300350761.
//
// The workaround is no longer needed with fetchGraphicBlock & C2Fence changes.
- // so we are reverting back to the logical sequence of the operations.
- if (android::media::codec::provider_->stop_hal_before_surface()) {
+ // so we are reverting back to the logical sequence of the operations when
+ // AIDL HALs are selected.
+ // When the HIDL HALs are selected, we retained workaround(the reversed
+ // order) as default in order to keep legacy behavior.
+ bool stopHalBeforeSurface =
+ Codec2Client::IsAidlSelected() ||
+ property_get_bool("debug.codec2.stop_hal_before_surface", false);
+ if (stopHalBeforeSurface && android::media::codec::provider_->stop_hal_before_surface()) {
comp->release();
mChannel->stopUseOutputSurface(pushBlankBuffer);
} else {
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 3984b83..c7ab82f 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -2784,7 +2784,16 @@
}
void CCodecBufferChannel::setInfoBuffer(const std::shared_ptr<C2InfoBuffer> &buffer) {
- mInfoBuffers.push_back(buffer);
+ if (mInputSurface == nullptr) {
+ mInfoBuffers.push_back(buffer);
+ } else {
+ std::list<std::unique_ptr<C2Work>> items;
+ std::unique_ptr<C2Work> work(new C2Work);
+ work->input.infoBuffers.emplace_back(*buffer);
+ work->worklets.emplace_back(new C2Worklet);
+ items.push_back(std::move(work));
+ c2_status_t err = mComponent->queue(&items);
+ }
}
status_t toStatusT(c2_status_t c2s, c2_operation_t c2op) {
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 2afe80c..d6b1163 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -1707,14 +1707,14 @@
mSelectedDeviceId = deviceId;
if (mStatus == NO_ERROR) {
if (isOffloadedOrDirect_l()) {
- if (mState == STATE_STOPPED || mState == STATE_FLUSHED) {
- ALOGD("%s(%d): creating a new AudioTrack", __func__, mPortId);
- result = restoreTrack_l("setOutputDevice", true /* forceRestore */);
- } else {
+ if (isPlaying_l()) {
ALOGW("%s(%d). Offloaded or Direct track is not STOPPED or FLUSHED. "
"State: %s.",
__func__, mPortId, stateToString(mState));
result = INVALID_OPERATION;
+ } else {
+ ALOGD("%s(%d): creating a new AudioTrack", __func__, mPortId);
+ result = restoreTrack_l("setOutputDevice", true /* forceRestore */);
}
} else {
// allow track invalidation when track is not playing to propagate
diff --git a/media/libaudiohal/impl/Android.bp b/media/libaudiohal/impl/Android.bp
index dd8f021..1a6b949 100644
--- a/media/libaudiohal/impl/Android.bp
+++ b/media/libaudiohal/impl/Android.bp
@@ -227,11 +227,11 @@
"latest_android_hardware_audio_core_sounddose_ndk_shared",
"latest_android_hardware_audio_effect_ndk_shared",
"latest_android_media_audio_common_types_ndk_shared",
+ "latest_av_audio_types_aidl_ndk_shared",
],
shared_libs: [
"android.hardware.common-V2-ndk",
"android.hardware.common.fmq-V1-ndk",
- "av-audio-types-aidl-V1-ndk",
"libaudio_aidl_conversion_common_cpp",
"libaudio_aidl_conversion_common_ndk",
"libaudio_aidl_conversion_common_ndk_cpp",
diff --git a/media/libaudiohal/impl/StreamHalAidl.cpp b/media/libaudiohal/impl/StreamHalAidl.cpp
index 4c0ba37..4f01ec7 100644
--- a/media/libaudiohal/impl/StreamHalAidl.cpp
+++ b/media/libaudiohal/impl/StreamHalAidl.cpp
@@ -200,8 +200,12 @@
StreamDescriptor::Reply reply;
switch (state) {
case StreamDescriptor::State::ACTIVE:
+ case StreamDescriptor::State::DRAINING:
+ case StreamDescriptor::State::TRANSFERRING:
RETURN_STATUS_IF_ERROR(pause(&reply));
- if (reply.state != StreamDescriptor::State::PAUSED) {
+ if (reply.state != StreamDescriptor::State::PAUSED &&
+ reply.state != StreamDescriptor::State::DRAIN_PAUSED &&
+ reply.state != StreamDescriptor::State::TRANSFER_PAUSED) {
ALOGE("%s: unexpected stream state: %s (expected PAUSED)",
__func__, toString(reply.state).c_str());
return INVALID_OPERATION;
@@ -209,6 +213,7 @@
FALLTHROUGH_INTENDED;
case StreamDescriptor::State::PAUSED:
case StreamDescriptor::State::DRAIN_PAUSED:
+ case StreamDescriptor::State::TRANSFER_PAUSED:
if (mIsInput) return flush();
RETURN_STATUS_IF_ERROR(flush(&reply));
if (reply.state != StreamDescriptor::State::IDLE) {
@@ -276,11 +281,12 @@
return OK;
}
-status_t StreamHalAidl::getObservablePosition(int64_t *frames, int64_t *timestamp) {
+status_t StreamHalAidl::getObservablePosition(int64_t* frames, int64_t* timestamp,
+ StatePositions* statePositions) {
ALOGV("%p %s::%s", this, getClassName().c_str(), __func__);
if (!mStream) return NO_INIT;
StreamDescriptor::Reply reply;
- RETURN_STATUS_IF_ERROR(updateCountersIfNeeded(&reply));
+ RETURN_STATUS_IF_ERROR(updateCountersIfNeeded(&reply, statePositions));
*frames = std::max<int64_t>(0, reply.observable.frames);
*timestamp = std::max<int64_t>(0, reply.observable.timeNs);
return OK;
@@ -437,8 +443,12 @@
if (auto state = getState(); state == StreamDescriptor::State::DRAINING) {
// Retrieve the current state together with position counters unconditionally
// to ensure that the state on our side gets updated.
- sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(),
- nullptr, true /*safeFromNonWorkerThread */);
+ sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(), nullptr,
+ true /*safeFromNonWorkerThread */);
+ // For compatibility with HIDL behavior, apply a "soft" position reset
+ // after receiving the "drain ready" callback.
+ std::lock_guard l(mLock);
+ mStatePositions.framesAtFlushOrDrain = mLastReply.observable.frames;
} else {
ALOGW("%s: unexpected onDrainReady in the state %s", __func__, toString(state).c_str());
}
@@ -505,9 +515,9 @@
}
status_t StreamHalAidl::sendCommand(
- const ::aidl::android::hardware::audio::core::StreamDescriptor::Command &command,
+ const ::aidl::android::hardware::audio::core::StreamDescriptor::Command& command,
::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply,
- bool safeFromNonWorkerThread) {
+ bool safeFromNonWorkerThread, StatePositions* statePositions) {
// TIME_CHECK(); // TODO(b/243839867) reenable only when optimized.
if (!safeFromNonWorkerThread) {
const pid_t workerTid = mWorkerTid.load(std::memory_order_acquire);
@@ -539,6 +549,23 @@
}
mLastReply = *reply;
mLastReplyExpirationNs = uptimeNanos() + mLastReplyLifeTimeNs;
+ if (!mIsInput && reply->status == STATUS_OK) {
+ if (command.getTag() == StreamDescriptor::Command::standby &&
+ reply->state == StreamDescriptor::State::STANDBY) {
+ mStatePositions.framesAtStandby = reply->observable.frames;
+ } else if (command.getTag() == StreamDescriptor::Command::flush &&
+ reply->state == StreamDescriptor::State::IDLE) {
+ mStatePositions.framesAtFlushOrDrain = reply->observable.frames;
+ } else if (!mContext.isAsynchronous() &&
+ command.getTag() == StreamDescriptor::Command::drain &&
+ (reply->state == StreamDescriptor::State::IDLE ||
+ reply->state == StreamDescriptor::State::DRAINING)) {
+ mStatePositions.framesAtFlushOrDrain = reply->observable.frames;
+ } // for asynchronous drain, the frame count is saved in 'onAsyncDrainReady'
+ }
+ if (statePositions != nullptr) {
+ *statePositions = mStatePositions;
+ }
}
}
switch (reply->status) {
@@ -554,7 +581,8 @@
}
status_t StreamHalAidl::updateCountersIfNeeded(
- ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply) {
+ ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply,
+ StatePositions* statePositions) {
bool doUpdate = false;
{
std::lock_guard l(mLock);
@@ -564,10 +592,13 @@
// Since updates are paced, it is OK to perform them from any thread, they should
// not interfere with I/O operations of the worker.
return sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(),
- reply, true /*safeFromNonWorkerThread */);
+ reply, true /*safeFromNonWorkerThread */, statePositions);
} else if (reply != nullptr) { // provide cached reply
std::lock_guard l(mLock);
*reply = mLastReply;
+ if (statePositions != nullptr) {
+ *statePositions = mStatePositions;
+ }
}
return OK;
}
@@ -654,21 +685,27 @@
return transfer(const_cast<void*>(buffer), bytes, written);
}
-status_t StreamOutHalAidl::getRenderPosition(uint32_t *dspFrames) {
+status_t StreamOutHalAidl::getRenderPosition(uint64_t *dspFrames) {
if (dspFrames == nullptr) {
return BAD_VALUE;
}
int64_t aidlFrames = 0, aidlTimestamp = 0;
- RETURN_STATUS_IF_ERROR(getObservablePosition(&aidlFrames, &aidlTimestamp));
- *dspFrames = static_cast<uint32_t>(aidlFrames);
+ StatePositions statePositions{};
+ RETURN_STATUS_IF_ERROR(
+ getObservablePosition(&aidlFrames, &aidlTimestamp, &statePositions));
+ // Number of audio frames since the stream has exited standby.
+ // See the table at the start of 'StreamHalInterface' on when it needs to reset.
+ int64_t mostRecentResetPoint;
+ if (!mContext.isAsynchronous() && audio_has_proportional_frames(mConfig.format)) {
+ mostRecentResetPoint = statePositions.framesAtStandby;
+ } else {
+ mostRecentResetPoint =
+ std::max(statePositions.framesAtStandby, statePositions.framesAtFlushOrDrain);
+ }
+ *dspFrames = aidlFrames <= mostRecentResetPoint ? 0 : aidlFrames - mostRecentResetPoint;
return OK;
}
-status_t StreamOutHalAidl::getNextWriteTimestamp(int64_t *timestamp __unused) {
- // Obsolete, use getPresentationPosition.
- return INVALID_OPERATION;
-}
-
status_t StreamOutHalAidl::setCallback(wp<StreamOutHalInterfaceCallback> callback) {
ALOGD("%p %s", this, __func__);
TIME_CHECK();
@@ -722,13 +759,26 @@
return BAD_VALUE;
}
int64_t aidlFrames = 0, aidlTimestamp = 0;
- RETURN_STATUS_IF_ERROR(getObservablePosition(&aidlFrames, &aidlTimestamp));
- *frames = aidlFrames;
+ StatePositions statePositions{};
+ RETURN_STATUS_IF_ERROR(getObservablePosition(&aidlFrames, &aidlTimestamp, &statePositions));
+ // See the table at the start of 'StreamHalInterface'.
+ if (!mContext.isAsynchronous() && audio_has_proportional_frames(mConfig.format)) {
+ *frames = aidlFrames;
+ } else {
+ const int64_t mostRecentResetPoint =
+ std::max(statePositions.framesAtStandby, statePositions.framesAtFlushOrDrain);
+ *frames = aidlFrames <= mostRecentResetPoint ? 0 : aidlFrames - mostRecentResetPoint;
+ }
timestamp->tv_sec = aidlTimestamp / NANOS_PER_SECOND;
timestamp->tv_nsec = aidlTimestamp - timestamp->tv_sec * NANOS_PER_SECOND;
return OK;
}
+status_t StreamOutHalAidl::presentationComplete() {
+ ALOGD("%p %s::%s", this, getClassName().c_str(), __func__);
+ return OK;
+}
+
status_t StreamOutHalAidl::updateSourceMetadata(
const StreamOutHalInterface::SourceMetadata& sourceMetadata) {
TIME_CHECK();
diff --git a/media/libaudiohal/impl/StreamHalAidl.h b/media/libaudiohal/impl/StreamHalAidl.h
index b20eb00..fff7a92 100644
--- a/media/libaudiohal/impl/StreamHalAidl.h
+++ b/media/libaudiohal/impl/StreamHalAidl.h
@@ -194,6 +194,11 @@
// For tests.
friend class sp<StreamHalAidl>;
+ struct StatePositions {
+ int64_t framesAtFlushOrDrain;
+ int64_t framesAtStandby;
+ };
+
template<class T>
static std::shared_ptr<::aidl::android::hardware::audio::core::IStreamCommon> getStreamCommon(
const std::shared_ptr<T>& stream);
@@ -212,7 +217,8 @@
status_t getLatency(uint32_t *latency);
// Always returns non-negative values.
- status_t getObservablePosition(int64_t *frames, int64_t *timestamp);
+ status_t getObservablePosition(int64_t* frames, int64_t* timestamp,
+ StatePositions* statePositions = nullptr);
// Always returns non-negative values.
status_t getHardwarePosition(int64_t *frames, int64_t *timestamp);
@@ -268,11 +274,13 @@
// Note: Since `sendCommand` takes mLock while holding mCommandReplyLock, never call
// it with `mLock` being held.
status_t sendCommand(
- const ::aidl::android::hardware::audio::core::StreamDescriptor::Command &command,
+ const ::aidl::android::hardware::audio::core::StreamDescriptor::Command& command,
::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply = nullptr,
- bool safeFromNonWorkerThread = false);
+ bool safeFromNonWorkerThread = false,
+ StatePositions* statePositions = nullptr);
status_t updateCountersIfNeeded(
- ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply = nullptr);
+ ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply = nullptr,
+ StatePositions* statePositions = nullptr);
const std::shared_ptr<::aidl::android::hardware::audio::core::IStreamCommon> mStream;
const std::shared_ptr<::aidl::android::media::audio::IHalAdapterVendorExtension> mVendorExt;
@@ -280,6 +288,9 @@
std::mutex mLock;
::aidl::android::hardware::audio::core::StreamDescriptor::Reply mLastReply GUARDED_BY(mLock);
int64_t mLastReplyExpirationNs GUARDED_BY(mLock) = 0;
+ // Cached values of observable positions when the stream last entered certain state.
+ // Updated for output streams only.
+ StatePositions mStatePositions GUARDED_BY(mLock) = {};
// mStreamPowerLog is used for audio signal power logging.
StreamPowerLog mStreamPowerLog;
std::atomic<pid_t> mWorkerTid = -1;
@@ -308,10 +319,7 @@
// Return the number of audio frames written by the audio dsp to DAC since
// the output has exited standby.
- status_t getRenderPosition(uint32_t *dspFrames) override;
-
- // Get the local time at which the next write to the audio driver will be presented.
- status_t getNextWriteTimestamp(int64_t *timestamp) override;
+ status_t getRenderPosition(uint64_t *dspFrames) override;
// Set the callback for notifying completion of non-blocking write and drain.
status_t setCallback(wp<StreamOutHalInterfaceCallback> callback) override;
@@ -331,12 +339,19 @@
// Requests notification when data buffered by the driver/hardware has been played.
status_t drain(bool earlyNotify) override;
- // Notifies to the audio driver to flush the queued data.
+ // Notifies to the audio driver to flush (that is, drop) the queued data. Stream must
+ // already be paused before calling 'flush'.
status_t flush() override;
// Return a recent count of the number of audio frames presented to an external observer.
+ // This excludes frames which have been written but are still in the pipeline. See the
+ // table at the start of the 'StreamOutHalInterface' for the specification of the frame
+ // count behavior w.r.t. 'flush', 'drain' and 'standby' operations.
status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp) override;
+ // Notifies the HAL layer that the framework considers the current playback as completed.
+ status_t presentationComplete() override;
+
// Called when the metadata of the stream's source has been changed.
status_t updateSourceMetadata(const SourceMetadata& sourceMetadata) override;
@@ -413,6 +428,7 @@
// Return a recent count of the number of audio frames received and
// the clock time associated with that frame count.
+ // The count must not reset to zero when a PCM input enters standby.
status_t getCapturePosition(int64_t *frames, int64_t *time) override;
// Get active microphones
diff --git a/media/libaudiohal/impl/StreamHalHidl.cpp b/media/libaudiohal/impl/StreamHalHidl.cpp
index 77c75db..9e22700 100644
--- a/media/libaudiohal/impl/StreamHalHidl.cpp
+++ b/media/libaudiohal/impl/StreamHalHidl.cpp
@@ -17,6 +17,8 @@
#define LOG_TAG "StreamHalHidl"
//#define LOG_NDEBUG 0
+#include <cinttypes>
+
#include <android/hidl/manager/1.0/IServiceManager.h>
#include <hwbinder/IPCThreadState.h>
#include <media/AudioParameter.h>
@@ -589,32 +591,39 @@
return OK;
}
-status_t StreamOutHalHidl::getRenderPosition(uint32_t *dspFrames) {
+status_t StreamOutHalHidl::getRenderPosition(uint64_t *dspFrames) {
// TIME_CHECK(); // TODO(b/243839867) reenable only when optimized.
if (mStream == 0) return NO_INIT;
Result retval;
+ uint32_t halPosition = 0;
Return<void> ret = mStream->getRenderPosition(
[&](Result r, uint32_t d) {
retval = r;
if (retval == Result::OK) {
- *dspFrames = d;
+ halPosition = d;
}
});
- return processReturn("getRenderPosition", ret, retval);
-}
+ status_t status = processReturn("getRenderPosition", ret, retval);
+ if (status != OK) {
+ return status;
+ }
+ // Maintain a 64-bit render position using the 32-bit result from the HAL.
+ // This delta calculation relies on the arithmetic overflow behavior
+ // of integers. For example (100 - 0xFFFFFFF0) = 116.
+ std::lock_guard l(mPositionMutex);
+ const auto truncatedPosition = (uint32_t)mRenderPosition;
+ int32_t deltaHalPosition; // initialization not needed, overwitten by __builtin_sub_overflow()
+ (void) __builtin_sub_overflow(halPosition, truncatedPosition, &deltaHalPosition);
-status_t StreamOutHalHidl::getNextWriteTimestamp(int64_t *timestamp) {
- TIME_CHECK();
- if (mStream == 0) return NO_INIT;
- Result retval;
- Return<void> ret = mStream->getNextWriteTimestamp(
- [&](Result r, int64_t t) {
- retval = r;
- if (retval == Result::OK) {
- *timestamp = t;
- }
- });
- return processReturn("getRenderPosition", ret, retval);
+ if (deltaHalPosition >= 0) {
+ mRenderPosition += deltaHalPosition;
+ } else if (mExpectRetrograde) {
+ mExpectRetrograde = false;
+ mRenderPosition -= static_cast<uint64_t>(-deltaHalPosition);
+ ALOGW("Retrograde motion of %" PRId32 " frames", -deltaHalPosition);
+ }
+ *dspFrames = mRenderPosition;
+ return OK;
}
status_t StreamOutHalHidl::setCallback(wp<StreamOutHalInterfaceCallback> callback) {
@@ -667,9 +676,23 @@
status_t StreamOutHalHidl::flush() {
TIME_CHECK();
if (mStream == 0) return NO_INIT;
+ {
+ std::lock_guard l(mPositionMutex);
+ mRenderPosition = 0;
+ mExpectRetrograde = false;
+ }
return processReturn("pause", mStream->flush());
}
+status_t StreamOutHalHidl::standby() {
+ {
+ std::lock_guard l(mPositionMutex);
+ mRenderPosition = 0;
+ mExpectRetrograde = false;
+ }
+ return StreamHalHidl::standby();
+}
+
status_t StreamOutHalHidl::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) {
// TIME_CHECK(); // TODO(b/243839867) reenable only when optimized.
if (mStream == 0) return NO_INIT;
@@ -696,6 +719,16 @@
}
}
+status_t StreamOutHalHidl::presentationComplete() {
+ // Avoid suppressing retrograde motion in mRenderPosition for gapless offload/direct when
+ // transitioning between tracks.
+ // The HAL resets the frame position without flush/stop being called, but calls back prior to
+ // this event. So, on the next occurrence of retrograde motion, we permit backwards movement of
+ // mRenderPosition.
+ mExpectRetrograde = true;
+ return OK;
+}
+
#if MAJOR_VERSION == 2
status_t StreamOutHalHidl::updateSourceMetadata(
const StreamOutHalInterface::SourceMetadata& /* sourceMetadata */) {
diff --git a/media/libaudiohal/impl/StreamHalHidl.h b/media/libaudiohal/impl/StreamHalHidl.h
index 48da633..433e0a3 100644
--- a/media/libaudiohal/impl/StreamHalHidl.h
+++ b/media/libaudiohal/impl/StreamHalHidl.h
@@ -18,10 +18,12 @@
#define ANDROID_HARDWARE_STREAM_HAL_HIDL_H
#include <atomic>
+#include <mutex>
#include PATH(android/hardware/audio/CORE_TYPES_FILE_VERSION/IStream.h)
#include PATH(android/hardware/audio/CORE_TYPES_FILE_VERSION/IStreamIn.h)
#include PATH(android/hardware/audio/FILE_VERSION/IStreamOut.h)
+#include <android-base/thread_annotations.h>
#include <fmq/EventFlag.h>
#include <fmq/MessageQueue.h>
#include <media/audiohal/EffectHalInterface.h>
@@ -119,6 +121,9 @@
class StreamOutHalHidl : public StreamOutHalInterface, public StreamHalHidl {
public:
+ // Put the audio hardware input/output into standby mode (from StreamHalInterface).
+ status_t standby() override;
+
// Return the frame size (number of bytes per sample) of a stream.
virtual status_t getFrameSize(size_t *size);
@@ -136,10 +141,7 @@
// Return the number of audio frames written by the audio dsp to DAC since
// the output has exited standby.
- virtual status_t getRenderPosition(uint32_t *dspFrames);
-
- // Get the local time at which the next write to the audio driver will be presented.
- virtual status_t getNextWriteTimestamp(int64_t *timestamp);
+ virtual status_t getRenderPosition(uint64_t *dspFrames);
// Set the callback for notifying completion of non-blocking write and drain.
virtual status_t setCallback(wp<StreamOutHalInterfaceCallback> callback);
@@ -159,12 +161,19 @@
// Requests notification when data buffered by the driver/hardware has been played.
virtual status_t drain(bool earlyNotify);
- // Notifies to the audio driver to flush the queued data.
+ // Notifies to the audio driver to flush (that is, drop) the queued data. Stream must
+ // already be paused before calling 'flush'.
virtual status_t flush();
// Return a recent count of the number of audio frames presented to an external observer.
+ // This excludes frames which have been written but are still in the pipeline. See the
+ // table at the start of the 'StreamOutHalInterface' for the specification of the frame
+ // count behavior w.r.t. 'flush', 'drain' and 'standby' operations.
virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
+ // Notifies the HAL layer that the framework considers the current playback as completed.
+ status_t presentationComplete() override;
+
// Called when the metadata of the stream's source has been changed.
status_t updateSourceMetadata(const SourceMetadata& sourceMetadata) override;
@@ -221,6 +230,10 @@
std::unique_ptr<StatusMQ> mStatusMQ;
std::atomic<pid_t> mWriterClient;
EventFlag* mEfGroup;
+ std::mutex mPositionMutex;
+ // Used to expand correctly the 32-bit position from the HAL.
+ uint64_t mRenderPosition GUARDED_BY(mPositionMutex) = 0;
+ bool mExpectRetrograde GUARDED_BY(mPositionMutex) = false; // See 'presentationComplete'.
// Can not be constructed directly by clients.
StreamOutHalHidl(const sp<::android::hardware::audio::CPP_VERSION::IStreamOut>& stream);
@@ -250,6 +263,7 @@
// Return a recent count of the number of audio frames received and
// the clock time associated with that frame count.
+ // The count must not reset to zero when a PCM input enters standby.
virtual status_t getCapturePosition(int64_t *frames, int64_t *time);
// Get active microphones
diff --git a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
index 37615af..585a895 100644
--- a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
@@ -135,6 +135,38 @@
virtual ~StreamOutHalInterfaceLatencyModeCallback() = default;
};
+/**
+ * On position reporting. There are two methods: 'getRenderPosition' and
+ * 'getPresentationPosition'. The first difference is that they may have a
+ * time offset because "render" position relates to what happens between
+ * ADSP and DAC, while "observable" position is relative to the external
+ * observer. The second difference is that 'getRenderPosition' always
+ * resets on standby (for all types of stream data) according to its
+ * definition. Since the original C definition of 'getRenderPosition' used
+ * 32-bit frame counters, and also because in complex playback chains that
+ * include wireless devices the "observable" position has more practical
+ * meaning, 'getRenderPosition' does not exist in the AIDL HAL interface.
+ * The table below summarizes frame count behavior for 'getPresentationPosition':
+ *
+ * | Mixed | Direct | Direct
+ * | | non-offload | offload
+ * ==============|============|==============|==============
+ * PCM and | Continuous | |
+ * encapsulated | | |
+ * bitstream | | |
+ * --------------|------------| Continuous† |
+ * Bitstream | | | Reset on
+ * encapsulated | | | flush, drain
+ * into PCM | | | and standby
+ * | Not | |
+ * --------------| supported |--------------|
+ * Bitstream | | Reset on |
+ * | | flush, drain |
+ * | | and standby |
+ * | | |
+ *
+ * † - on standby, reset of the frame count happens at the framework level.
+ */
class StreamOutHalInterface : public virtual StreamHalInterface {
public:
// Return the audio hardware driver estimated latency in milliseconds.
@@ -151,10 +183,7 @@
// Return the number of audio frames written by the audio dsp to DAC since
// the output has exited standby.
- virtual status_t getRenderPosition(uint32_t *dspFrames) = 0;
-
- // Get the local time at which the next write to the audio driver will be presented.
- virtual status_t getNextWriteTimestamp(int64_t *timestamp) = 0;
+ virtual status_t getRenderPosition(uint64_t *dspFrames) = 0;
// Set the callback for notifying completion of non-blocking write and drain.
// The callback must be owned by someone else. The output stream does not own it
@@ -176,12 +205,19 @@
// Requests notification when data buffered by the driver/hardware has been played.
virtual status_t drain(bool earlyNotify) = 0;
- // Notifies to the audio driver to flush the queued data.
+ // Notifies to the audio driver to flush (that is, drop) the queued data. Stream must
+ // already be paused before calling 'flush'.
virtual status_t flush() = 0;
// Return a recent count of the number of audio frames presented to an external observer.
+ // This excludes frames which have been written but are still in the pipeline. See the
+ // table at the start of the 'StreamOutHalInterface' for the specification of the frame
+ // count behavior w.r.t. 'flush', 'drain' and 'standby' operations.
virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp) = 0;
+ // Notifies the HAL layer that the framework considers the current playback as completed.
+ virtual status_t presentationComplete() = 0;
+
struct SourceMetadata {
std::vector<playback_track_metadata_v7_t> tracks;
};
@@ -270,6 +306,7 @@
// Return a recent count of the number of audio frames received and
// the clock time associated with that frame count.
+ // The count must not reset to zero when a PCM input enters standby.
virtual status_t getCapturePosition(int64_t *frames, int64_t *time) = 0;
// Get active microphones
diff --git a/media/libstagefright/VideoRenderQualityTracker.cpp b/media/libstagefright/VideoRenderQualityTracker.cpp
index eb9ac0f..bf29b1d 100644
--- a/media/libstagefright/VideoRenderQualityTracker.cpp
+++ b/media/libstagefright/VideoRenderQualityTracker.cpp
@@ -302,13 +302,6 @@
mRenderDurationMs += (actualRenderTimeUs - mLastRenderTimeUs) / 1000;
}
- // Now that a frame has been rendered, the previously skipped frames can be processed as skipped
- // frames since the app is not skipping them to terminate playback.
- for (int64_t contentTimeUs : mPendingSkippedFrameContentTimeUsList) {
- processMetricsForSkippedFrame(contentTimeUs);
- }
- mPendingSkippedFrameContentTimeUsList = {};
-
// We can render a pending queued frame if it's the last frame of the video, so release it
// immediately.
if (contentTimeUs == mTunnelFrameQueuedContentTimeUs && mTunnelFrameQueuedContentTimeUs != -1) {
@@ -332,9 +325,25 @@
(long long) contentTimeUs, (long long) nextExpectedFrame.contentTimeUs);
break;
}
+ // Process all skipped frames before the dropped frame.
+ while (!mPendingSkippedFrameContentTimeUsList.empty()) {
+ if (mPendingSkippedFrameContentTimeUsList.front() >= nextExpectedFrame.contentTimeUs) {
+ break;
+ }
+ processMetricsForSkippedFrame(mPendingSkippedFrameContentTimeUsList.front());
+ mPendingSkippedFrameContentTimeUsList.pop_front();
+ }
processMetricsForDroppedFrame(nextExpectedFrame.contentTimeUs,
nextExpectedFrame.desiredRenderTimeUs);
}
+ // Process all skipped frames before the rendered frame.
+ while (!mPendingSkippedFrameContentTimeUsList.empty()) {
+ if (mPendingSkippedFrameContentTimeUsList.front() >= nextExpectedFrame.contentTimeUs) {
+ break;
+ }
+ processMetricsForSkippedFrame(mPendingSkippedFrameContentTimeUsList.front());
+ mPendingSkippedFrameContentTimeUsList.pop_front();
+ }
processMetricsForRenderedFrame(nextExpectedFrame.contentTimeUs,
nextExpectedFrame.desiredRenderTimeUs, actualRenderTimeUs,
freezeEventOut, judderEventOut);
diff --git a/media/libstagefright/rtsp/fuzzer/Android.bp b/media/libstagefright/rtsp/fuzzer/Android.bp
index a2791ba..ff64af5 100644
--- a/media/libstagefright/rtsp/fuzzer/Android.bp
+++ b/media/libstagefright/rtsp/fuzzer/Android.bp
@@ -29,11 +29,19 @@
header_libs: [
"libstagefright_rtsp_headers",
],
- fuzz_config:{
+ fuzz_config: {
cc: [
- "android-media-fuzzing-reports@google.com",
+ "android-media-playback@google.com",
],
componentid: 155276,
+ hotlists: [
+ "4593311",
+ ],
+ description: "This fuzzer targets the APIs of libstagefright_rtsp",
+ vector: "local_privileges_required",
+ service_privilege: "privileged",
+ users: "multi_user",
+ fuzzed_code_usage: "shipped",
},
}
@@ -44,7 +52,7 @@
],
defaults: [
"libstagefright_rtsp_fuzzer_defaults",
- ]
+ ],
}
cc_fuzz {
@@ -55,7 +63,7 @@
defaults: [
"libstagefright_rtsp_fuzzer_defaults",
],
- shared_libs:[
+ shared_libs: [
"libandroid_net",
"libbase",
"libstagefright",
diff --git a/media/libstagefright/timedtext/test/fuzzer/Android.bp b/media/libstagefright/timedtext/test/fuzzer/Android.bp
index 6590ebb..8724d51 100644
--- a/media/libstagefright/timedtext/test/fuzzer/Android.bp
+++ b/media/libstagefright/timedtext/test/fuzzer/Android.bp
@@ -48,8 +48,16 @@
],
fuzz_config: {
cc: [
- "android-media-fuzzing-reports@google.com",
+ "android-media-playback@google.com",
],
- componentid: 155276,
+ componentid: 42195,
+ hotlists: [
+ "4593311",
+ ],
+ description: "This fuzzer targets the APIs of libstagefright_timedtext",
+ vector: "local_no_privileges_required",
+ service_privilege: "constrained",
+ users: "multi_user",
+ fuzzed_code_usage: "shipped",
},
}
diff --git a/media/module/extractors/mp4/MPEG4Extractor.cpp b/media/module/extractors/mp4/MPEG4Extractor.cpp
index b3707c8..cb2994e 100644
--- a/media/module/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/module/extractors/mp4/MPEG4Extractor.cpp
@@ -1615,39 +1615,6 @@
mLastTrack->timescale = ntohl(timescale);
- // 14496-12 says all ones means indeterminate, but some files seem to use
- // 0 instead. We treat both the same.
- int64_t duration = 0;
- if (version == 1) {
- if (mDataSource->readAt(
- timescale_offset + 4, &duration, sizeof(duration))
- < (ssize_t)sizeof(duration)) {
- return ERROR_IO;
- }
- if (duration != -1) {
- duration = ntoh64(duration);
- }
- } else {
- uint32_t duration32;
- if (mDataSource->readAt(
- timescale_offset + 4, &duration32, sizeof(duration32))
- < (ssize_t)sizeof(duration32)) {
- return ERROR_IO;
- }
- if (duration32 != 0xffffffff) {
- duration = ntohl(duration32);
- }
- }
- if (duration != 0 && mLastTrack->timescale != 0) {
- long double durationUs = ((long double)duration * 1000000) / mLastTrack->timescale;
- if (durationUs < 0 || durationUs > INT64_MAX) {
- ALOGE("cannot represent %lld * 1000000 / %lld in 64 bits",
- (long long) duration, (long long) mLastTrack->timescale);
- return ERROR_MALFORMED;
- }
- AMediaFormat_setInt64(mLastTrack->meta, AMEDIAFORMAT_KEY_DURATION, durationUs);
- }
-
uint8_t lang[2];
off64_t lang_offset;
if (version == 1) {
@@ -3907,17 +3874,18 @@
}
int32_t id;
+ int64_t duration;
if (version == 1) {
// we can get ctime value from U64_AT(&buffer[4])
// we can get mtime value from U64_AT(&buffer[12])
id = U32_AT(&buffer[20]);
- // we can get duration value from U64_AT(&buffer[28])
+ duration = U64_AT(&buffer[28]);
} else if (version == 0) {
// we can get ctime value from U32_AT(&buffer[4])
// we can get mtime value from U32_AT(&buffer[8])
id = U32_AT(&buffer[12]);
- // we can get duration value from U32_AT(&buffer[20])
+ duration = U32_AT(&buffer[20]);
} else {
return ERROR_UNSUPPORTED;
}
@@ -3926,6 +3894,15 @@
return ERROR_MALFORMED;
AMediaFormat_setInt32(mLastTrack->meta, AMEDIAFORMAT_KEY_TRACK_ID, id);
+ if (duration != 0 && mHeaderTimescale != 0) {
+ long double durationUs = ((long double)duration * 1000000) / mHeaderTimescale;
+ if (durationUs < 0 || durationUs > INT64_MAX) {
+ ALOGE("cannot represent %lld * 1000000 / %lld in 64 bits",
+ (long long) duration, (long long) mHeaderTimescale);
+ return ERROR_MALFORMED;
+ }
+ AMediaFormat_setInt64(mLastTrack->meta, AMEDIAFORMAT_KEY_DURATION, durationUs);
+ }
size_t matrixOffset = dynSize + 16;
int32_t a00 = U32_AT(&buffer[matrixOffset]);
diff --git a/media/ndk/Android.bp b/media/ndk/Android.bp
index 9ec7700..3d873df 100644
--- a/media/ndk/Android.bp
+++ b/media/ndk/Android.bp
@@ -192,7 +192,6 @@
header_libs: [
"libstagefright_headers",
"libmedia_headers",
- "libstagefright_headers",
],
shared_libs: [
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 1d7c356..60abb58 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2690,14 +2690,17 @@
}
}
- // Set DIRECT flag if current thread is DirectOutputThread. This can
- // happen when the playback is rerouted to direct output thread by
+ // Set DIRECT/OFFLOAD flag if current thread is DirectOutputThread/OffloadThread.
+ // This can happen when the playback is rerouted to direct output/offload thread by
// dynamic audio policy.
// Do NOT report the flag changes back to client, since the client
- // doesn't explicitly request a direct flag.
+ // doesn't explicitly request a direct/offload flag.
audio_output_flags_t trackFlags = *flags;
if (mType == DIRECT) {
trackFlags = static_cast<audio_output_flags_t>(trackFlags | AUDIO_OUTPUT_FLAG_DIRECT);
+ } else if (mType == OFFLOAD) {
+ trackFlags = static_cast<audio_output_flags_t>(trackFlags |
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT);
}
*afTrackFlags = trackFlags;
@@ -3357,9 +3360,9 @@
return NO_ERROR;
} else {
status_t status;
- uint32_t frames;
+ uint64_t frames = 0;
status = mOutput->getRenderPosition(&frames);
- *dspFrames = (size_t)frames;
+ *dspFrames = (uint32_t)frames;
return status;
}
}
@@ -5903,7 +5906,7 @@
vaf = v * sendLevel * (1. / MAX_GAIN_INT);
}
- track->setFinalVolume(vrf, vlf);
+ track->setFinalVolume(vlf, vrf);
// Delegate volume control to effect in track effect chain if needed
if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
diff --git a/services/audioflinger/datapath/AudioStreamIn.cpp b/services/audioflinger/datapath/AudioStreamIn.cpp
index 76618f4..165ac25 100644
--- a/services/audioflinger/datapath/AudioStreamIn.cpp
+++ b/services/audioflinger/datapath/AudioStreamIn.cpp
@@ -58,7 +58,7 @@
if (mHalFormatHasProportionalFrames &&
(flags & AUDIO_INPUT_FLAG_DIRECT) == AUDIO_INPUT_FLAG_DIRECT) {
- // For DirectRecord reset timestamp to 0 on standby.
+ // For DirectRecord reset position to 0 on standby.
const uint64_t adjustedPosition = (halPosition <= mFramesReadAtStandby) ?
0 : (halPosition - mFramesReadAtStandby);
// Scale from HAL sample rate to application rate.
diff --git a/services/audioflinger/datapath/AudioStreamOut.cpp b/services/audioflinger/datapath/AudioStreamOut.cpp
index 9851f3a..a686ff6 100644
--- a/services/audioflinger/datapath/AudioStreamOut.cpp
+++ b/services/audioflinger/datapath/AudioStreamOut.cpp
@@ -51,42 +51,17 @@
return NO_INIT;
}
- uint32_t halPosition = 0;
+ uint64_t halPosition = 0;
const status_t status = stream->getRenderPosition(&halPosition);
if (status != NO_ERROR) {
return status;
}
-
- // Maintain a 64-bit render position using the 32-bit result from the HAL.
- // This delta calculation relies on the arithmetic overflow behavior
- // of integers. For example (100 - 0xFFFFFFF0) = 116.
- const auto truncatedPosition = (uint32_t)mRenderPosition;
- int32_t deltaHalPosition; // initialization not needed, overwitten by __builtin_sub_overflow()
- (void) __builtin_sub_overflow(halPosition, truncatedPosition, &deltaHalPosition);
-
- if (deltaHalPosition > 0) {
- mRenderPosition += deltaHalPosition;
- } else if (mExpectRetrograde) {
- mExpectRetrograde = false;
- mRenderPosition -= static_cast<uint64_t>(-deltaHalPosition);
- }
// Scale from HAL sample rate to application rate.
- *frames = mRenderPosition / mRateMultiplier;
+ *frames = halPosition / mRateMultiplier;
return status;
}
-// return bottom 32-bits of the render position
-status_t AudioStreamOut::getRenderPosition(uint32_t *frames)
-{
- uint64_t position64 = 0;
- const status_t status = getRenderPosition(&position64);
- if (status == NO_ERROR) {
- *frames = (uint32_t)position64;
- }
- return status;
-}
-
status_t AudioStreamOut::getPresentationPosition(uint64_t *frames, struct timespec *timestamp)
{
if (stream == nullptr) {
@@ -101,7 +76,7 @@
if (mHalFormatHasProportionalFrames &&
(flags & AUDIO_OUTPUT_FLAG_DIRECT) == AUDIO_OUTPUT_FLAG_DIRECT) {
- // For DirectTrack reset timestamp to 0 on standby.
+ // For DirectTrack reset position to 0 on standby.
const uint64_t adjustedPosition = (halPosition <= mFramesWrittenAtStandby) ?
0 : (halPosition - mFramesWrittenAtStandby);
// Scale from HAL sample rate to application rate.
@@ -179,8 +154,6 @@
int AudioStreamOut::flush()
{
- mRenderPosition = 0;
- mExpectRetrograde = false;
mFramesWritten = 0;
mFramesWrittenAtStandby = 0;
const status_t result = stream->flush();
@@ -189,12 +162,14 @@
int AudioStreamOut::standby()
{
- mRenderPosition = 0;
- mExpectRetrograde = false;
mFramesWrittenAtStandby = mFramesWritten;
return stream->standby();
}
+void AudioStreamOut::presentationComplete() {
+ stream->presentationComplete();
+}
+
ssize_t AudioStreamOut::write(const void *buffer, size_t numBytes)
{
size_t bytesWritten;
diff --git a/services/audioflinger/datapath/AudioStreamOut.h b/services/audioflinger/datapath/AudioStreamOut.h
index ea41bba..2c9fb3e 100644
--- a/services/audioflinger/datapath/AudioStreamOut.h
+++ b/services/audioflinger/datapath/AudioStreamOut.h
@@ -51,9 +51,6 @@
virtual ~AudioStreamOut();
- // Get the bottom 32-bits of the 64-bit render position.
- status_t getRenderPosition(uint32_t *frames);
-
virtual status_t getRenderPosition(uint64_t *frames);
virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
@@ -91,21 +88,14 @@
virtual status_t flush();
virtual status_t standby();
- // Avoid suppressing retrograde motion in mRenderPosition for gapless offload/direct when
- // transitioning between tracks.
- // The HAL resets the frame position without flush/stop being called, but calls back prior to
- // this event. So, on the next occurrence of retrograde motion, we permit backwards movement of
- // mRenderPosition.
- virtual void presentationComplete() { mExpectRetrograde = true; }
+ virtual void presentationComplete();
protected:
uint64_t mFramesWritten = 0; // reset by flush
uint64_t mFramesWrittenAtStandby = 0;
- uint64_t mRenderPosition = 0; // reset by flush, standby, or presentation complete
int mRateMultiplier = 1;
bool mHalFormatHasProportionalFrames = false;
size_t mHalFrameSize = 0;
- bool mExpectRetrograde = false; // see presentationComplete
};
} // namespace android
diff --git a/services/audioparameterparser/Android.bp b/services/audioparameterparser/Android.bp
index f5feece..1c1c1e1 100644
--- a/services/audioparameterparser/Android.bp
+++ b/services/audioparameterparser/Android.bp
@@ -35,10 +35,10 @@
name: "android.hardware.audio.parameter_parser.example_defaults",
defaults: [
"latest_android_hardware_audio_core_ndk_shared",
+ "latest_av_audio_types_aidl_ndk_shared",
],
shared_libs: [
- "av-audio-types-aidl-V1-ndk",
"libbase",
"libbinder_ndk",
],
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index d027564..747af4a 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -561,6 +561,7 @@
audio_port_config config = {};
devicePort->toAudioPortConfig(&config);
config.config_mask = AUDIO_PORT_CONFIG_GAIN;
+ config.gain.mode = gains[0]->getMode();
config.gain.values[0] = gainValueMb;
return mClientInterface->setAudioPortConfig(&config, 0) == NO_ERROR;
}