Merge "MediaCodec: add methods to query/subscribe vendor parameters"
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 9cabd8b..200e92d 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -182,6 +182,7 @@
// This is set by AudioTrack.setBufferSizeInFrames().
// A write will not fill the buffer above this limit.
volatile uint32_t mBufferSizeInFrames; // effective size of the buffer
+ volatile uint32_t mStartThresholdInFrames; // min frames in buffer to start streaming
public:
@@ -216,6 +217,8 @@
};
size_t frameCount() const { return mFrameCount; }
+ uint32_t getStartThresholdInFrames() const;
+ uint32_t setStartThresholdInFrames(uint32_t startThresholdInFrames);
protected:
// These refer to shared memory, and are virtual addresses with respect to the current process.
diff --git a/media/codec2/components/aac/C2SoftAacDec.cpp b/media/codec2/components/aac/C2SoftAacDec.cpp
index 3e6b0ff..332696d 100644
--- a/media/codec2/components/aac/C2SoftAacDec.cpp
+++ b/media/codec2/components/aac/C2SoftAacDec.cpp
@@ -55,6 +55,8 @@
namespace android {
constexpr char COMPONENT_NAME[] = "c2.android.aac.decoder";
+constexpr size_t kDefaultOutputPortDelay = 2;
+constexpr size_t kMaxOutputPortDelay = 16;
class C2SoftAacDec::IntfImpl : public SimpleInterface<void>::BaseParams {
public:
@@ -73,7 +75,9 @@
addParameter(
DefineParam(mActualOutputDelay, C2_PARAMKEY_OUTPUT_DELAY)
- .withConstValue(new C2PortActualDelayTuning::output(2u))
+ .withDefault(new C2PortActualDelayTuning::output(kDefaultOutputPortDelay))
+ .withFields({C2F(mActualOutputDelay, value).inRange(0, kMaxOutputPortDelay)})
+ .withSetter(Setter<decltype(*mActualOutputDelay)>::StrictValueWithNoDeps)
.build());
addParameter(
@@ -263,6 +267,7 @@
mAACDecoder(nullptr),
mStreamInfo(nullptr),
mSignalledError(false),
+ mOutputPortDelay(kDefaultOutputPortDelay),
mOutputDelayRingBuffer(nullptr) {
}
@@ -915,6 +920,29 @@
int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
+ size_t numSamplesInOutput = mStreamInfo->frameSize * mStreamInfo->numChannels;
+ if (numSamplesInOutput > 0) {
+ size_t actualOutputPortDelay = (outputDelay + numSamplesInOutput - 1) / numSamplesInOutput;
+ if (actualOutputPortDelay > mOutputPortDelay) {
+ mOutputPortDelay = actualOutputPortDelay;
+ ALOGV("New Output port delay %zu ", mOutputPortDelay);
+
+ C2PortActualDelayTuning::output outputPortDelay(mOutputPortDelay);
+ std::vector<std::unique_ptr<C2SettingResult>> failures;
+ c2_status_t err =
+ mIntf->config({&outputPortDelay}, C2_MAY_BLOCK, &failures);
+ if (err == OK) {
+ work->worklets.front()->output.configUpdate.push_back(
+ C2Param::Copy(outputPortDelay));
+ } else {
+ ALOGE("Cannot set output delay");
+ mSignalledError = true;
+ work->workletsProcessed = 1u;
+ work->result = C2_CORRUPTED;
+ return;
+ }
+ }
+ }
mBuffersInfo.push_back(std::move(inInfo));
work->workletsProcessed = 0u;
if (!eos && mOutputDelayCompensated < outputDelay) {
diff --git a/media/codec2/components/aac/C2SoftAacDec.h b/media/codec2/components/aac/C2SoftAacDec.h
index 965c29e..986187c 100644
--- a/media/codec2/components/aac/C2SoftAacDec.h
+++ b/media/codec2/components/aac/C2SoftAacDec.h
@@ -57,6 +57,7 @@
size_t mInputBufferCount;
size_t mOutputBufferCount;
bool mSignalledError;
+ size_t mOutputPortDelay;
struct Info {
uint64_t frameIndex;
size_t bufferSize;
diff --git a/media/codec2/sfplugin/C2OMXNode.cpp b/media/codec2/sfplugin/C2OMXNode.cpp
index ab73245..2460490 100644
--- a/media/codec2/sfplugin/C2OMXNode.cpp
+++ b/media/codec2/sfplugin/C2OMXNode.cpp
@@ -33,12 +33,14 @@
#include <OMX_IndexExt.h>
#include <android/fdsan.h>
+#include <media/stagefright/foundation/ColorUtils.h>
#include <media/stagefright/omx/OMXUtils.h>
#include <media/stagefright/MediaErrors.h>
#include <ui/Fence.h>
#include <ui/GraphicBuffer.h>
#include <utils/Thread.h>
+#include "utils/Codec2Mapper.h"
#include "C2OMXNode.h"
namespace android {
@@ -71,6 +73,23 @@
jobs->cond.broadcast();
}
+ void setDataspace(android_dataspace dataspace) {
+ Mutexed<Jobs>::Locked jobs(mJobs);
+ ColorUtils::convertDataSpaceToV0(dataspace);
+ jobs->configUpdate.emplace_back(new C2StreamDataSpaceInfo::input(0u, dataspace));
+ int32_t standard;
+ int32_t transfer;
+ int32_t range;
+ ColorUtils::getColorConfigFromDataSpace(dataspace, &range, &standard, &transfer);
+ std::unique_ptr<C2StreamColorAspectsInfo::input> colorAspects =
+ std::make_unique<C2StreamColorAspectsInfo::input>(0u);
+ if (C2Mapper::map(standard, &colorAspects->primaries, &colorAspects->matrix)
+ && C2Mapper::map(transfer, &colorAspects->transfer)
+ && C2Mapper::map(range, &colorAspects->range)) {
+ jobs->configUpdate.push_back(std::move(colorAspects));
+ }
+ }
+
protected:
bool threadLoop() override {
constexpr nsecs_t kIntervalNs = nsecs_t(10) * 1000 * 1000; // 10ms
@@ -102,6 +121,9 @@
uniqueFds.push_back(std::move(queue.workList.front().fd1));
queue.workList.pop_front();
}
+ for (const std::unique_ptr<C2Param> ¶m : jobs->configUpdate) {
+ items.front()->input.configUpdate.emplace_back(C2Param::Copy(*param));
+ }
jobs.unlock();
for (int fenceFd : fenceFds) {
@@ -119,6 +141,7 @@
queued = true;
}
if (queued) {
+ jobs->configUpdate.clear();
return true;
}
if (i == 0) {
@@ -161,6 +184,7 @@
std::map<std::weak_ptr<Codec2Client::Component>,
Queue,
std::owner_less<std::weak_ptr<Codec2Client::Component>>> queues;
+ std::vector<std::unique_ptr<C2Param>> configUpdate;
Condition cond;
};
Mutexed<Jobs> mJobs;
@@ -172,6 +196,9 @@
mQueueThread(new QueueThread) {
android_fdsan_set_error_level(ANDROID_FDSAN_ERROR_LEVEL_WARN_ALWAYS);
mQueueThread->run("C2OMXNode", PRIORITY_AUDIO);
+
+ Mutexed<android_dataspace>::Locked ds(mDataspace);
+ *ds = HAL_DATASPACE_UNKNOWN;
}
status_t C2OMXNode::freeNode() {
@@ -459,8 +486,11 @@
android_dataspace dataSpace = (android_dataspace)msg.u.event_data.data1;
uint32_t pixelFormat = msg.u.event_data.data3;
- // TODO: set dataspace on component to see if it impacts color aspects
ALOGD("dataspace changed to %#x pixel format: %#x", dataSpace, pixelFormat);
+ mQueueThread->setDataspace(dataSpace);
+
+ Mutexed<android_dataspace>::Locked ds(mDataspace);
+ *ds = dataSpace;
return OK;
}
@@ -493,4 +523,8 @@
(void)mBufferSource->onInputBufferEmptied(bufferId, -1);
}
+android_dataspace C2OMXNode::getDataspace() {
+ return *mDataspace.lock();
+}
+
} // namespace android
diff --git a/media/codec2/sfplugin/C2OMXNode.h b/media/codec2/sfplugin/C2OMXNode.h
index 1717c96..9c04969 100644
--- a/media/codec2/sfplugin/C2OMXNode.h
+++ b/media/codec2/sfplugin/C2OMXNode.h
@@ -93,6 +93,11 @@
*/
void onInputBufferDone(c2_cntr64_t index);
+ /**
+ * Returns dataspace information from GraphicBufferSource.
+ */
+ android_dataspace getDataspace();
+
private:
std::weak_ptr<Codec2Client::Component> mComp;
sp<IOMXBufferSource> mBufferSource;
@@ -101,6 +106,7 @@
uint32_t mWidth;
uint32_t mHeight;
uint64_t mUsage;
+ Mutexed<android_dataspace> mDataspace;
// WORKAROUND: timestamp adjustment
diff --git a/media/codec2/sfplugin/CCodec.cpp b/media/codec2/sfplugin/CCodec.cpp
index d61185d..dc7dbe5 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -211,8 +211,6 @@
(OMX_INDEXTYPE)OMX_IndexParamConsumerUsageBits,
&usage, sizeof(usage));
- // NOTE: we do not use/pass through color aspects from GraphicBufferSource as we
- // communicate that directly to the component.
mSource->configure(
mOmxNode, static_cast<hardware::graphics::common::V1_0::Dataspace>(mDataSpace));
return OK;
@@ -411,6 +409,10 @@
mNode->onInputBufferDone(index);
}
+ android_dataspace getDataspace() override {
+ return mNode->getDataspace();
+ }
+
private:
sp<HGraphicBufferSource> mSource;
sp<C2OMXNode> mNode;
@@ -1571,6 +1573,7 @@
outputFormat = config->mOutputFormat = config->mOutputFormat->dup();
if (config->mInputSurface) {
err2 = config->mInputSurface->start();
+ config->mInputSurfaceDataspace = config->mInputSurface->getDataspace();
}
buffersBoundToCodec = config->mBuffersBoundToCodec;
}
@@ -1658,6 +1661,7 @@
if (config->mInputSurface) {
config->mInputSurface->disconnect();
config->mInputSurface = nullptr;
+ config->mInputSurfaceDataspace = HAL_DATASPACE_UNKNOWN;
}
}
{
@@ -1707,6 +1711,7 @@
if (config->mInputSurface) {
config->mInputSurface->disconnect();
config->mInputSurface = nullptr;
+ config->mInputSurfaceDataspace = HAL_DATASPACE_UNKNOWN;
}
}
diff --git a/media/codec2/sfplugin/CCodecConfig.cpp b/media/codec2/sfplugin/CCodecConfig.cpp
index 31fcc72..ad28545 100644
--- a/media/codec2/sfplugin/CCodecConfig.cpp
+++ b/media/codec2/sfplugin/CCodecConfig.cpp
@@ -1129,6 +1129,12 @@
insertion.first->second = std::move(p);
}
}
+ if (mInputSurface
+ && (domain & mOutputDomain)
+ && mInputSurfaceDataspace != mInputSurface->getDataspace()) {
+ changed = true;
+ mInputSurfaceDataspace = mInputSurface->getDataspace();
+ }
ALOGV("updated configuration has %zu params (%s)", mCurrentConfig.size(),
changed ? "CHANGED" : "no change");
@@ -1357,7 +1363,6 @@
msg->removeEntryAt(msg->findEntryByName("color-matrix"));
}
-
// calculate dataspace for raw graphic buffers if not specified by component, or if
// using surface with unspecified aspects (as those must be defaulted which may change
// the dataspace)
@@ -1395,6 +1400,23 @@
}
}
+ if (mInputSurface) {
+ android_dataspace dataspace = mInputSurface->getDataspace();
+ ColorUtils::convertDataSpaceToV0(dataspace);
+ int32_t standard;
+ ColorUtils::getColorConfigFromDataSpace(dataspace, &range, &standard, &transfer);
+ if (range != 0) {
+ msg->setInt32(KEY_COLOR_RANGE, range);
+ }
+ if (standard != 0) {
+ msg->setInt32(KEY_COLOR_STANDARD, standard);
+ }
+ if (transfer != 0) {
+ msg->setInt32(KEY_COLOR_TRANSFER, transfer);
+ }
+ msg->setInt32("android._dataspace", dataspace);
+ }
+
// HDR static info
C2HdrStaticMetadataStruct hdr;
diff --git a/media/codec2/sfplugin/CCodecConfig.h b/media/codec2/sfplugin/CCodecConfig.h
index 2e16bc9..417b773 100644
--- a/media/codec2/sfplugin/CCodecConfig.h
+++ b/media/codec2/sfplugin/CCodecConfig.h
@@ -125,6 +125,7 @@
std::shared_ptr<InputSurfaceWrapper> mInputSurface;
std::unique_ptr<InputSurfaceWrapper::Config> mISConfig;
+ android_dataspace mInputSurfaceDataspace;
/// the current configuration. Updated after configure() and based on configUpdate in
/// onWorkDone
diff --git a/media/codec2/sfplugin/InputSurfaceWrapper.h b/media/codec2/sfplugin/InputSurfaceWrapper.h
index bb35763..3ddae01 100644
--- a/media/codec2/sfplugin/InputSurfaceWrapper.h
+++ b/media/codec2/sfplugin/InputSurfaceWrapper.h
@@ -106,6 +106,11 @@
*/
virtual void onInputBufferDone(c2_cntr64_t /* index */) {}
+ /**
+ * Returns dataspace information from GraphicBufferSource.
+ */
+ virtual android_dataspace getDataspace() { return mDataSpace; }
+
protected:
android_dataspace mDataSpace;
};
diff --git a/media/codec2/sfplugin/tests/CCodecConfig_test.cpp b/media/codec2/sfplugin/tests/CCodecConfig_test.cpp
index c9caa01..7c660dc 100644
--- a/media/codec2/sfplugin/tests/CCodecConfig_test.cpp
+++ b/media/codec2/sfplugin/tests/CCodecConfig_test.cpp
@@ -208,6 +208,24 @@
.withSetter(Setter<C2StreamPixelAspectRatioInfo::output>)
.build());
+ if (isEncoder) {
+ addParameter(
+ DefineParam(mInputBitrate, C2_PARAMKEY_BITRATE)
+ .withDefault(new C2StreamBitrateInfo::input(0u))
+ .withFields({C2F(mInputBitrate, value).any()})
+ .withSetter(Setter<C2StreamBitrateInfo::input>)
+ .build());
+
+ addParameter(
+ DefineParam(mOutputBitrate, C2_PARAMKEY_BITRATE)
+ .withDefault(new C2StreamBitrateInfo::output(0u))
+ .withFields({C2F(mOutputBitrate, value).any()})
+ .calculatedAs(
+ Copy<C2StreamBitrateInfo::output, C2StreamBitrateInfo::input>,
+ mInputBitrate)
+ .build());
+ }
+
// TODO: more SDK params
}
private:
@@ -221,11 +239,19 @@
std::shared_ptr<C2StreamVendorInt64Info::output> mInt64Output;
std::shared_ptr<C2PortVendorStringInfo::input> mStringInput;
std::shared_ptr<C2StreamPixelAspectRatioInfo::output> mPixelAspectRatio;
+ std::shared_ptr<C2StreamBitrateInfo::input> mInputBitrate;
+ std::shared_ptr<C2StreamBitrateInfo::output> mOutputBitrate;
template<typename T>
static C2R Setter(bool, C2P<T> &) {
return C2R::Ok();
}
+
+ template<typename ME, typename DEP>
+ static C2R Copy(bool, C2P<ME> &me, const C2P<DEP> &dep) {
+ me.set().value = dep.v.value;
+ return C2R::Ok();
+ }
};
Impl mImpl;
@@ -457,4 +483,97 @@
<< "mInputFormat = " << mConfig.mInputFormat->debugString().c_str();
}
+TEST_F(CCodecConfigTest, DataspaceUpdate) {
+ init(C2Component::DOMAIN_VIDEO, C2Component::KIND_ENCODER, MIMETYPE_VIDEO_AVC);
+
+ ASSERT_EQ(OK, mConfig.initialize(mReflector, mConfigurable));
+ class InputSurfaceStub : public InputSurfaceWrapper {
+ public:
+ ~InputSurfaceStub() override = default;
+ status_t connect(const std::shared_ptr<Codec2Client::Component> &) override {
+ return OK;
+ }
+ void disconnect() override {}
+ status_t start() override { return OK; }
+ status_t signalEndOfInputStream() override { return OK; }
+ status_t configure(Config &) override { return OK; }
+ };
+ mConfig.mInputSurface = std::make_shared<InputSurfaceStub>();
+
+ sp<AMessage> format{new AMessage};
+ format->setInt32(KEY_COLOR_RANGE, COLOR_RANGE_LIMITED);
+ format->setInt32(KEY_COLOR_STANDARD, COLOR_STANDARD_BT709);
+ format->setInt32(KEY_COLOR_TRANSFER, COLOR_TRANSFER_SDR_VIDEO);
+ format->setInt32(KEY_BIT_RATE, 100);
+
+ std::vector<std::unique_ptr<C2Param>> configUpdate;
+ ASSERT_EQ(OK, mConfig.getConfigUpdateFromSdkParams(
+ mConfigurable, format, D::ALL, C2_MAY_BLOCK, &configUpdate));
+ ASSERT_TRUE(mConfig.updateConfiguration(configUpdate, D::ALL));
+
+ int32_t range{0};
+ ASSERT_TRUE(mConfig.mOutputFormat->findInt32(KEY_COLOR_RANGE, &range))
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+ EXPECT_EQ(COLOR_RANGE_LIMITED, range)
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+
+ int32_t standard{0};
+ ASSERT_TRUE(mConfig.mOutputFormat->findInt32(KEY_COLOR_STANDARD, &standard))
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+ EXPECT_EQ(COLOR_STANDARD_BT709, standard)
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+
+ int32_t transfer{0};
+ ASSERT_TRUE(mConfig.mOutputFormat->findInt32(KEY_COLOR_TRANSFER, &transfer))
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+ EXPECT_EQ(COLOR_TRANSFER_SDR_VIDEO, transfer)
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+
+ mConfig.mInputSurface->setDataSpace(HAL_DATASPACE_BT2020_PQ);
+
+ // Dataspace from input surface should override the configured setting
+ mConfig.updateFormats(D::ALL);
+
+ ASSERT_TRUE(mConfig.mOutputFormat->findInt32(KEY_COLOR_RANGE, &range))
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+ EXPECT_EQ(COLOR_RANGE_FULL, range)
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+
+ ASSERT_TRUE(mConfig.mOutputFormat->findInt32(KEY_COLOR_STANDARD, &standard))
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+ EXPECT_EQ(COLOR_STANDARD_BT2020, standard)
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+
+ ASSERT_TRUE(mConfig.mOutputFormat->findInt32(KEY_COLOR_TRANSFER, &transfer))
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+ EXPECT_EQ(COLOR_TRANSFER_ST2084, transfer)
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+
+ // Simulate bitrate update
+ format = new AMessage;
+ format->setInt32(KEY_BIT_RATE, 200);
+ configUpdate.clear();
+ ASSERT_EQ(OK, mConfig.getConfigUpdateFromSdkParams(
+ mConfigurable, format, D::ALL, C2_MAY_BLOCK, &configUpdate));
+ ASSERT_EQ(OK, mConfig.setParameters(mConfigurable, configUpdate, C2_MAY_BLOCK));
+
+ // Color information should remain the same
+ mConfig.updateFormats(D::ALL);
+
+ ASSERT_TRUE(mConfig.mOutputFormat->findInt32(KEY_COLOR_RANGE, &range))
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+ EXPECT_EQ(COLOR_RANGE_FULL, range)
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+
+ ASSERT_TRUE(mConfig.mOutputFormat->findInt32(KEY_COLOR_STANDARD, &standard))
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+ EXPECT_EQ(COLOR_STANDARD_BT2020, standard)
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+
+ ASSERT_TRUE(mConfig.mOutputFormat->findInt32(KEY_COLOR_TRANSFER, &transfer))
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+ EXPECT_EQ(COLOR_TRANSFER_ST2084, transfer)
+ << "mOutputFormat = " << mConfig.mOutputFormat->debugString().c_str();
+}
+
} // namespace android
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 19d1d1a..e37cc12 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -1249,6 +1249,46 @@
return finalBufferSize;
}
+ssize_t AudioTrack::getStartThresholdInFrames() const
+{
+ AutoMutex lock(mLock);
+ if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
+ return NO_INIT;
+ }
+ return (ssize_t) mProxy->getStartThresholdInFrames();
+}
+
+ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
+{
+ if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
+ // contractually we could simply return the current threshold in frames
+ // to indicate the request was ignored, but we return an error here.
+ return BAD_VALUE;
+ }
+ AutoMutex lock(mLock);
+ // We do not permit calling setStartThresholdInFrames() between the AudioTrack
+ // default ctor AudioTrack() and set(...) but rather fail such an attempt.
+ // (To do so would require a cached mOrigStartThresholdInFrames and we may
+ // not have proper validation for the actual set value).
+ if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
+ return NO_INIT;
+ }
+ const uint32_t original = mProxy->getStartThresholdInFrames();
+ const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
+ if (original != final) {
+ android::mediametrics::LogItem(mMetricsId)
+ .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
+ .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
+ .record();
+ if (original > final) {
+ // restart track if it was disabled by audioflinger due to previous underrun
+ // and we reduced the number of frames for the threshold.
+ restartIfDisabled();
+ }
+ }
+ return final;
+}
+
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
@@ -2562,6 +2602,10 @@
staticPosition = mStaticProxy->getPosition().unsignedValue();
}
+ // save the old startThreshold and framecount
+ const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
+ const uint32_t originalFrameCount = mProxy->frameCount();
+
// See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
// causes a lot of churn on the service side, and it can reject starting
// playback of a previously created track. May also apply to other cases.
@@ -2616,6 +2660,18 @@
return mAudioTrack->applyVolumeShaper(shaper.mConfiguration, operationToEnd);
});
+ // restore the original start threshold if different than frameCount.
+ if (originalStartThresholdInFrames != originalFrameCount) {
+ // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
+ // and does not trigger a restart.
+ // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
+ // Any start would be triggered on the mState == ACTIVE check below.
+ const uint32_t currentThreshold =
+ mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
+ ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
+ "%s(%d) startThresholdInFrames changing from %u to %u",
+ __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
+ }
if (mState == STATE_ACTIVE) {
result = mAudioTrack->start();
}
diff --git a/media/libaudioclient/AudioTrackShared.cpp b/media/libaudioclient/AudioTrackShared.cpp
index f1f8f9c..35719be 100644
--- a/media/libaudioclient/AudioTrackShared.cpp
+++ b/media/libaudioclient/AudioTrackShared.cpp
@@ -17,6 +17,7 @@
#define LOG_TAG "AudioTrackShared"
//#define LOG_NDEBUG 0
+#include <atomic>
#include <android-base/macros.h>
#include <private/media/AudioTrackShared.h>
#include <utils/Log.h>
@@ -33,6 +34,21 @@
return sizeof(T) > sizeof(size_t) && x > (T) SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x;
}
+// compile-time safe atomics. TODO: update all methods to use it
+template <typename T>
+T android_atomic_load(const volatile T* addr) {
+ static_assert(sizeof(T) == sizeof(std::atomic<T>)); // no extra sync data required.
+ static_assert(std::atomic<T>::is_always_lock_free); // no hash lock somewhere.
+ return atomic_load((std::atomic<T>*)addr); // memory_order_seq_cst
+}
+
+template <typename T>
+void android_atomic_store(const volatile T* addr, T value) {
+ static_assert(sizeof(T) == sizeof(std::atomic<T>)); // no extra sync data required.
+ static_assert(std::atomic<T>::is_always_lock_free); // no hash lock somewhere.
+ atomic_store((std::atomic<T>*)addr, value); // memory_order_seq_cst
+}
+
// incrementSequence is used to determine the next sequence value
// for the loop and position sequence counters. It should return
// a value between "other" + 1 and "other" + INT32_MAX, the choice of
@@ -51,6 +67,7 @@
: mServer(0), mFutex(0), mMinimum(0)
, mVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY), mSampleRate(0), mSendLevel(0)
, mBufferSizeInFrames(0)
+ , mStartThresholdInFrames(0) // filled in by the server.
, mFlags(0)
{
memset(&u, 0, sizeof(u));
@@ -66,6 +83,26 @@
{
}
+uint32_t Proxy::getStartThresholdInFrames() const
+{
+ const uint32_t startThresholdInFrames =
+ android_atomic_load(&mCblk->mStartThresholdInFrames);
+ if (startThresholdInFrames == 0 || startThresholdInFrames > mFrameCount) {
+ ALOGD("%s: startThresholdInFrames %u not between 1 and frameCount %zu, "
+ "setting to frameCount",
+ __func__, startThresholdInFrames, mFrameCount);
+ return mFrameCount;
+ }
+ return startThresholdInFrames;
+}
+
+uint32_t Proxy::setStartThresholdInFrames(uint32_t startThresholdInFrames)
+{
+ const uint32_t actual = std::min((size_t)startThresholdInFrames, frameCount());
+ android_atomic_store(&mCblk->mStartThresholdInFrames, actual);
+ return actual;
+}
+
// ---------------------------------------------------------------------------
ClientProxy::ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
@@ -663,6 +700,7 @@
, mTimestampMutator(&cblk->mExtendedTimestampQueue)
{
cblk->mBufferSizeInFrames = frameCount;
+ cblk->mStartThresholdInFrames = frameCount;
}
__attribute__((no_sanitize("integer")))
@@ -900,11 +938,8 @@
}
audio_track_cblk_t* cblk = mCblk;
- int32_t flush = cblk->u.mStreaming.mFlush;
- if (flush != mFlush) {
- // FIXME should return an accurate value, but over-estimate is better than under-estimate
- return mFrameCount;
- }
+ flushBufferIfNeeded();
+
const int32_t rear = getRear();
ssize_t filled = audio_utils::safe_sub_overflow(rear, cblk->u.mStreaming.mFront);
// pipe should not already be overfull
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index fac4c83..4afa9c9 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -429,6 +429,19 @@
*/
ssize_t setBufferSizeInFrames(size_t size);
+ /* Returns the start threshold on the buffer for audio streaming
+ * or a negative value if the AudioTrack is not initialized.
+ */
+ ssize_t getStartThresholdInFrames() const;
+
+ /* Sets the start threshold in frames on the buffer for audio streaming.
+ *
+ * May be clamped internally. Returns the actual value set, or a negative
+ * value if the AudioTrack is not initialized or if the input
+ * is zero or greater than INT_MAX.
+ */
+ ssize_t setStartThresholdInFrames(size_t startThresholdInFrames);
+
/* Return the static buffer specified in constructor or set(), or 0 for streaming mode */
sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index da16477..0d5fe59 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -246,6 +246,10 @@
return status;
}
CoreUtils::AudioInputFlags hidlFlags;
+#if MAJOR_VERSION <= 5
+ // Some flags were specific to framework and must not leak to the HAL.
+ flags = static_cast<audio_input_flags_t>(flags & ~AUDIO_INPUT_FLAG_DIRECT);
+#endif
if (status_t status = CoreUtils::audioInputFlagsFromHal(flags, &hidlFlags); status != OK) {
return status;
}
@@ -278,10 +282,6 @@
sinkMetadata.tracks[0].destination.device(std::move(hidlOutputDevice));
}
#endif
-#if MAJOR_VERSION <= 5
- // Some flags were specific to framework and must not leak to the HAL.
- flags = static_cast<audio_input_flags_t>(flags & ~AUDIO_INPUT_FLAG_DIRECT);
-#endif
Return<void> ret = mDevice->openInputStream(
handle, hidlDevice, hidlConfig, hidlFlags, sinkMetadata,
[&](Result r, const sp<IStreamIn>& result, const AudioConfig& suggestedConfig) {
diff --git a/media/libeffects/preprocessing/Android.bp b/media/libeffects/preprocessing/Android.bp
index 87ed8b6..c6e036a 100644
--- a/media/libeffects/preprocessing/Android.bp
+++ b/media/libeffects/preprocessing/Android.bp
@@ -18,15 +18,10 @@
],
}
-cc_library {
- name: "libaudiopreprocessing",
+cc_defaults {
+ name: "libaudiopreprocessing-defaults",
vendor: true,
- relative_install_path: "soundfx",
host_supported: true,
- srcs: ["PreProcessing.cpp"],
- local_include_dirs: [
- ".",
- ],
cflags: [
"-Wall",
"-Werror",
@@ -46,7 +41,6 @@
header_libs: [
"libaudioeffects",
"libhardware_headers",
- "libwebrtc_absl_headers",
],
target: {
darwin: {
@@ -54,3 +48,13 @@
},
},
}
+
+cc_library {
+ name: "libaudiopreprocessing",
+ defaults: ["libaudiopreprocessing-defaults"],
+ relative_install_path: "soundfx",
+ srcs: ["PreProcessing.cpp"],
+ header_libs: [
+ "libwebrtc_absl_headers",
+ ],
+}
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index 3b0b6d6..19a8b2f 100644
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -105,9 +105,8 @@
webrtc::AudioProcessing* apm; // handle on webRTC audio processing module (APM)
// Audio Processing module builder
webrtc::AudioProcessingBuilder ap_builder;
- size_t apmFrameCount; // buffer size for webRTC process (10 ms)
- uint32_t apmSamplingRate; // webRTC APM sampling rate (8/16 or 32 kHz)
- size_t frameCount; // buffer size before input resampler ( <=> apmFrameCount)
+ // frameCount represents the size of the buffers used for processing, and must represent 10ms.
+ size_t frameCount;
uint32_t samplingRate; // sampling rate at effect process interface
uint32_t inChannelCount; // input channel count
uint32_t outChannelCount; // output channel count
@@ -119,21 +118,12 @@
webrtc::AudioProcessing::Config config;
webrtc::StreamConfig inputConfig; // input stream configuration
webrtc::StreamConfig outputConfig; // output stream configuration
- int16_t* inBuf; // input buffer used when resampling
- size_t inBufSize; // input buffer size in frames
- size_t framesIn; // number of frames in input buffer
- int16_t* outBuf; // output buffer used when resampling
- size_t outBufSize; // output buffer size in frames
- size_t framesOut; // number of frames in output buffer
uint32_t revChannelCount; // number of channels on reverse stream
uint32_t revEnabledMsk; // bit field containing IDs of enabled pre processors
// with reverse channel
uint32_t revProcessedMsk; // bit field containing IDs of pre processors with reverse
// channel already processed in current round
webrtc::StreamConfig revConfig; // reverse stream configuration.
- int16_t* revBuf; // reverse channel input buffer
- size_t revBufSize; // reverse channel input buffer size
- size_t framesRev; // number of frames in reverse channel input buffer
};
#ifdef DUAL_MIC_TEST
@@ -862,9 +852,7 @@
ALOGW("Session_CreateEffect could not get apm engine");
goto error;
}
- session->apmSamplingRate = kPreprocDefaultSr;
- session->apmFrameCount = (kPreprocDefaultSr) / 100;
- session->frameCount = session->apmFrameCount;
+ session->frameCount = kPreprocDefaultSr / 100;
session->samplingRate = kPreprocDefaultSr;
session->inChannelCount = kPreProcDefaultCnl;
session->outChannelCount = kPreProcDefaultCnl;
@@ -879,12 +867,6 @@
session->processedMsk = 0;
session->revEnabledMsk = 0;
session->revProcessedMsk = 0;
- session->inBuf = NULL;
- session->inBufSize = 0;
- session->outBuf = NULL;
- session->outBufSize = 0;
- session->revBuf = NULL;
- session->revBufSize = 0;
}
status = Effect_Create(&session->effects[procId], session, interface);
if (status < 0) {
@@ -908,13 +890,6 @@
if (session->createdMsk == 0) {
delete session->apm;
session->apm = NULL;
- delete session->inBuf;
- session->inBuf = NULL;
- free(session->outBuf);
- session->outBuf = NULL;
- delete session->revBuf;
- session->revBuf = NULL;
-
session->id = 0;
}
@@ -934,24 +909,8 @@
ALOGV("Session_SetConfig sr %d cnl %08x", config->inputCfg.samplingRate,
config->inputCfg.channels);
- // AEC implementation is limited to 16kHz
- if (config->inputCfg.samplingRate >= 32000 && !(session->createdMsk & (1 << PREPROC_AEC))) {
- session->apmSamplingRate = 32000;
- } else if (config->inputCfg.samplingRate >= 16000) {
- session->apmSamplingRate = 16000;
- } else if (config->inputCfg.samplingRate >= 8000) {
- session->apmSamplingRate = 8000;
- }
-
-
session->samplingRate = config->inputCfg.samplingRate;
- session->apmFrameCount = session->apmSamplingRate / 100;
- if (session->samplingRate == session->apmSamplingRate) {
- session->frameCount = session->apmFrameCount;
- } else {
- session->frameCount =
- (session->apmFrameCount * session->samplingRate) / session->apmSamplingRate;
- }
+ session->frameCount = session->samplingRate / 100;
session->inChannelCount = inCnl;
session->outChannelCount = outCnl;
session->inputConfig.set_sample_rate_hz(session->samplingRate);
@@ -963,13 +922,6 @@
session->revConfig.set_sample_rate_hz(session->samplingRate);
session->revConfig.set_num_channels(inCnl);
- // force process buffer reallocation
- session->inBufSize = 0;
- session->outBufSize = 0;
- session->framesIn = 0;
- session->framesOut = 0;
-
-
session->state = PREPROC_SESSION_STATE_CONFIG;
return 0;
}
@@ -1004,9 +956,6 @@
}
uint32_t inCnl = audio_channel_count_from_out_mask(config->inputCfg.channels);
session->revChannelCount = inCnl;
- // force process buffer reallocation
- session->revBufSize = 0;
- session->framesRev = 0;
return 0;
}
@@ -1023,12 +972,8 @@
void Session_SetProcEnabled(preproc_session_t* session, uint32_t procId, bool enabled) {
if (enabled) {
- if (session->enabledMsk == 0) {
- session->framesIn = 0;
- }
session->enabledMsk |= (1 << procId);
if (HasReverseStream(procId)) {
- session->framesRev = 0;
session->revEnabledMsk |= (1 << procId);
}
} else {
@@ -1117,43 +1062,24 @@
return -EINVAL;
}
+ if (inBuffer->frameCount != outBuffer->frameCount) {
+ ALOGW("inBuffer->frameCount %zu is not equal to outBuffer->frameCount %zu",
+ inBuffer->frameCount, outBuffer->frameCount);
+ return -EINVAL;
+ }
+
+ if (inBuffer->frameCount != session->frameCount) {
+ ALOGW("inBuffer->frameCount %zu != %zu representing 10ms at sampling rate %d",
+ inBuffer->frameCount, session->frameCount, session->samplingRate);
+ return -EINVAL;
+ }
+
session->processedMsk |= (1 << effect->procId);
// ALOGV("PreProcessingFx_Process In %d frames enabledMsk %08x processedMsk %08x",
// inBuffer->frameCount, session->enabledMsk, session->processedMsk);
-
if ((session->processedMsk & session->enabledMsk) == session->enabledMsk) {
effect->session->processedMsk = 0;
- size_t framesRq = outBuffer->frameCount;
- size_t framesWr = 0;
- if (session->framesOut) {
- size_t fr = session->framesOut;
- if (outBuffer->frameCount < fr) {
- fr = outBuffer->frameCount;
- }
- memcpy(outBuffer->s16, session->outBuf,
- fr * session->outChannelCount * sizeof(int16_t));
- memmove(session->outBuf, session->outBuf + fr * session->outChannelCount,
- (session->framesOut - fr) * session->outChannelCount * sizeof(int16_t));
- session->framesOut -= fr;
- framesWr += fr;
- }
- outBuffer->frameCount = framesWr;
- if (framesWr == framesRq) {
- inBuffer->frameCount = 0;
- return 0;
- }
-
- size_t fr = session->frameCount - session->framesIn;
- if (inBuffer->frameCount < fr) {
- fr = inBuffer->frameCount;
- }
- session->framesIn += fr;
- inBuffer->frameCount = fr;
- if (session->framesIn < session->frameCount) {
- return 0;
- }
- session->framesIn = 0;
if (int status = effect->session->apm->ProcessStream(
(const int16_t* const)inBuffer->s16,
(const webrtc::StreamConfig)effect->session->inputConfig,
@@ -1163,34 +1089,6 @@
ALOGE("Process Stream failed with error %d\n", status);
return status;
}
- outBuffer->frameCount = inBuffer->frameCount;
-
- if (session->outBufSize < session->framesOut + session->frameCount) {
- int16_t* buf;
- session->outBufSize = session->framesOut + session->frameCount;
- buf = (int16_t*)realloc(
- session->outBuf,
- session->outBufSize * session->outChannelCount * sizeof(int16_t));
- if (buf == NULL) {
- session->framesOut = 0;
- free(session->outBuf);
- session->outBuf = NULL;
- return -ENOMEM;
- }
- session->outBuf = buf;
- }
-
- fr = session->framesOut;
- if (framesRq - framesWr < fr) {
- fr = framesRq - framesWr;
- }
- memcpy(outBuffer->s16 + framesWr * session->outChannelCount, session->outBuf,
- fr * session->outChannelCount * sizeof(int16_t));
- memmove(session->outBuf, session->outBuf + fr * session->outChannelCount,
- (session->framesOut - fr) * session->outChannelCount * sizeof(int16_t));
- session->framesOut -= fr;
- outBuffer->frameCount += fr;
-
return 0;
} else {
return -ENODATA;
@@ -1565,6 +1463,18 @@
return -EINVAL;
}
+ if (inBuffer->frameCount != outBuffer->frameCount) {
+ ALOGW("inBuffer->frameCount %zu is not equal to outBuffer->frameCount %zu",
+ inBuffer->frameCount, outBuffer->frameCount);
+ return -EINVAL;
+ }
+
+ if (inBuffer->frameCount != session->frameCount) {
+ ALOGW("inBuffer->frameCount %zu != %zu representing 10ms at sampling rate %d",
+ inBuffer->frameCount, session->frameCount, session->samplingRate);
+ return -EINVAL;
+ }
+
session->revProcessedMsk |= (1 << effect->procId);
// ALOGV("PreProcessingFx_ProcessReverse In %d frames revEnabledMsk %08x revProcessedMsk
@@ -1573,16 +1483,6 @@
if ((session->revProcessedMsk & session->revEnabledMsk) == session->revEnabledMsk) {
effect->session->revProcessedMsk = 0;
- size_t fr = session->frameCount - session->framesRev;
- if (inBuffer->frameCount < fr) {
- fr = inBuffer->frameCount;
- }
- session->framesRev += fr;
- inBuffer->frameCount = fr;
- if (session->framesRev < session->frameCount) {
- return 0;
- }
- session->framesRev = 0;
if (int status = effect->session->apm->ProcessReverseStream(
(const int16_t* const)inBuffer->s16,
(const webrtc::StreamConfig)effect->session->revConfig,
diff --git a/media/libeffects/preprocessing/README.md b/media/libeffects/preprocessing/README.md
new file mode 100644
index 0000000..af46376
--- /dev/null
+++ b/media/libeffects/preprocessing/README.md
@@ -0,0 +1,7 @@
+# Preprocessing effects
+
+## Limitations
+- Preprocessing effects currently work on 10ms worth of data and do not support
+ arbitrary frame counts. This limiation comes from the underlying effects in
+ webrtc modules
+- There is currently no api to communicate this requirement
diff --git a/media/libeffects/preprocessing/benchmarks/Android.bp b/media/libeffects/preprocessing/benchmarks/Android.bp
index c1b2295..fbbcab4 100644
--- a/media/libeffects/preprocessing/benchmarks/Android.bp
+++ b/media/libeffects/preprocessing/benchmarks/Android.bp
@@ -11,27 +11,10 @@
cc_benchmark {
name: "preprocessing_benchmark",
- vendor: true,
+ defaults: ["libaudiopreprocessing-defaults"],
srcs: ["preprocessing_benchmark.cpp"],
- shared_libs: [
- "libaudioutils",
- "liblog",
- "libutils",
- ],
static_libs: [
"libaudiopreprocessing",
- "webrtc_audio_processing",
- ],
- cflags: [
- "-DWEBRTC_POSIX",
- "-fvisibility=default",
- "-Wall",
- "-Werror",
- "-Wextra",
- ],
- header_libs: [
- "libaudioeffects",
- "libhardware_headers",
- "libwebrtc_absl_headers",
+ "libaudioutils",
],
}
diff --git a/media/libeffects/preprocessing/tests/Android.bp b/media/libeffects/preprocessing/tests/Android.bp
index 18c6c98..d80b135 100644
--- a/media/libeffects/preprocessing/tests/Android.bp
+++ b/media/libeffects/preprocessing/tests/Android.bp
@@ -12,9 +12,8 @@
cc_test {
name: "EffectPreprocessingTest",
- vendor: true,
+ defaults: ["libaudiopreprocessing-defaults"],
gtest: true,
- host_supported: true,
test_suites: ["device-tests"],
srcs: [
"EffectPreprocessingTest.cpp",
@@ -23,46 +22,18 @@
static_libs: [
"libaudiopreprocessing",
"libaudioutils",
- "webrtc_audio_processing",
],
- shared_libs: [
- "liblog",
- ],
- header_libs: [
- "libaudioeffects",
- "libhardware_headers",
- ],
- target: {
- darwin: {
- enabled: false,
- },
- },
}
cc_test {
name: "AudioPreProcessingTest",
- vendor: true,
- host_supported: true,
+ defaults: ["libaudiopreprocessing-defaults"],
gtest: false,
srcs: ["PreProcessingTest.cpp"],
- shared_libs: [
- "libaudioutils",
- "liblog",
- "libutils",
- ],
static_libs: [
"libaudiopreprocessing",
- "webrtc_audio_processing",
+ "libaudioutils",
],
- header_libs: [
- "libaudioeffects",
- "libhardware_headers",
- ],
- target: {
- darwin: {
- enabled: false,
- },
- },
}
cc_test {
diff --git a/media/libeffects/preprocessing/tests/EffectTestHelper.h b/media/libeffects/preprocessing/tests/EffectTestHelper.h
index db06823..117cf7b 100644
--- a/media/libeffects/preprocessing/tests/EffectTestHelper.h
+++ b/media/libeffects/preprocessing/tests/EffectTestHelper.h
@@ -88,7 +88,8 @@
static constexpr size_t kNumChMasks = std::size(kChMasks);
- static constexpr size_t kSampleRates[] = {8000, 16000, 24000, 32000, 48000};
+ static constexpr size_t kSampleRates[] = {8000, 11025, 12000, 16000, 22050,
+ 24000, 32000, 44100, 48000};
static constexpr size_t kNumSampleRates = std::size(kSampleRates);
diff --git a/media/libeffects/preprocessing/tests/PreProcessingTest.cpp b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
index e0025fe..3bd93f8 100644
--- a/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
+++ b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
@@ -451,8 +451,8 @@
}
audio_buffer_t inputBuffer, outputBuffer;
audio_buffer_t farInBuffer{};
- inputBuffer.frameCount = samplesRead;
- outputBuffer.frameCount = samplesRead;
+ inputBuffer.frameCount = frameLength;
+ outputBuffer.frameCount = frameLength;
inputBuffer.s16 = in.data();
outputBuffer.s16 = out.data();
@@ -472,7 +472,7 @@
}
}
- farInBuffer.frameCount = samplesRead;
+ farInBuffer.frameCount = frameLength;
farInBuffer.s16 = farIn.data();
}
@@ -519,6 +519,7 @@
}
frameCounter += frameLength;
}
+ printf("frameCounter: [%d]\n", frameCounter);
// Release all the effect handles created
for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(effectHandle[i]);
diff --git a/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh b/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh
index 942f2ec..35da13e 100755
--- a/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh
+++ b/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh
@@ -59,9 +59,13 @@
fs_arr=(
8000
+ 11025
+ 12000
16000
+ 22050
24000
32000
+ 44100
48000
)
diff --git a/media/libeffects/proxy/EffectProxy.cpp b/media/libeffects/proxy/EffectProxy.cpp
index c010d68..be9f8c0 100644
--- a/media/libeffects/proxy/EffectProxy.cpp
+++ b/media/libeffects/proxy/EffectProxy.cpp
@@ -116,6 +116,16 @@
pContext->sube[SUB_FX_OFFLOAD] = sube[1];
pContext->desc[SUB_FX_OFFLOAD] = desc[1];
pContext->aeli[SUB_FX_OFFLOAD] = aeli[1];
+ } else {
+ ALOGE("Both effects have (or don't have) EFFECT_FLAG_HW_ACC_TUNNEL flag");
+ delete[] sube;
+ delete[] desc;
+ delete[] aeli;
+ delete[] pContext->sube;
+ delete[] pContext->desc;
+ delete[] pContext->aeli;
+ delete pContext;
+ return -EINVAL;
}
delete[] desc;
delete[] aeli;
diff --git a/media/libmediametrics/include/MediaMetricsConstants.h b/media/libmediametrics/include/MediaMetricsConstants.h
index 84388c9..674df17 100644
--- a/media/libmediametrics/include/MediaMetricsConstants.h
+++ b/media/libmediametrics/include/MediaMetricsConstants.h
@@ -139,6 +139,7 @@
#define AMEDIAMETRICS_PROP_SESSIONID "sessionId" // int32
#define AMEDIAMETRICS_PROP_SHARINGMODE "sharingMode" // string value, "exclusive", shared"
#define AMEDIAMETRICS_PROP_SOURCE "source" // string (AudioAttributes)
+#define AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES "startThresholdFrames" // int32 (AudioTrack)
#define AMEDIAMETRICS_PROP_STARTUPMS "startupMs" // double value
// State is "ACTIVE" or "STOPPED" for AudioRecord
#define AMEDIAMETRICS_PROP_STATE "state" // string
@@ -181,6 +182,7 @@
#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETMODE "setMode" // AudioFlinger
#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE "setBufferSize" // AudioTrack
#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM "setPlaybackParam" // AudioTrack
+#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD "setStartThreshold" // AudioTrack
#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOICEVOLUME "setVoiceVolume" // AudioFlinger
#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME "setVolume" // AudioTrack
#define AMEDIAMETRICS_PROP_EVENT_VALUE_START "start" // AudioTrack, AudioRecord
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 71a4ad8..224ec8b 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -5683,15 +5683,18 @@
int32_t range, standard, transfer;
convertCodecColorAspectsToPlatformAspects(aspects, &range, &standard, &transfer);
+ int32_t dsRange, dsStandard, dsTransfer;
+ getColorConfigFromDataSpace(dataSpace, &dsRange, &dsStandard, &dsTransfer);
+
// if some aspects are unspecified, use dataspace fields
if (range == 0) {
- range = (dataSpace & HAL_DATASPACE_RANGE_MASK) >> HAL_DATASPACE_RANGE_SHIFT;
+ range = dsRange;
}
if (standard == 0) {
- standard = (dataSpace & HAL_DATASPACE_STANDARD_MASK) >> HAL_DATASPACE_STANDARD_SHIFT;
+ standard = dsStandard;
}
if (transfer == 0) {
- transfer = (dataSpace & HAL_DATASPACE_TRANSFER_MASK) >> HAL_DATASPACE_TRANSFER_SHIFT;
+ transfer = dsTransfer;
}
mOutputFormat = mOutputFormat->dup(); // trigger an output format changed event
diff --git a/media/libstagefright/MediaCodecSource.cpp b/media/libstagefright/MediaCodecSource.cpp
index 0f7df24..876d06c 100644
--- a/media/libstagefright/MediaCodecSource.cpp
+++ b/media/libstagefright/MediaCodecSource.cpp
@@ -775,12 +775,9 @@
android_dataspace dataspace = static_cast<android_dataspace>(ds);
ColorUtils::convertDataSpaceToV0(dataspace);
ALOGD("Updating dataspace to %x", dataspace);
- int32_t standard = (int32_t(dataspace) & HAL_DATASPACE_STANDARD_MASK)
- >> HAL_DATASPACE_STANDARD_SHIFT;
- int32_t transfer = (int32_t(dataspace) & HAL_DATASPACE_TRANSFER_MASK)
- >> HAL_DATASPACE_TRANSFER_SHIFT;
- int32_t range = (int32_t(dataspace) & HAL_DATASPACE_RANGE_MASK)
- >> HAL_DATASPACE_RANGE_SHIFT;
+ int32_t standard, transfer, range;
+ ColorUtils::getColorConfigFromDataSpace(
+ dataspace, &range, &standard, &transfer);
sp<AMessage> msg = new AMessage;
msg->setInt32(KEY_COLOR_STANDARD, standard);
msg->setInt32(KEY_COLOR_TRANSFER, transfer);
diff --git a/media/libstagefright/foundation/ColorUtils.cpp b/media/libstagefright/foundation/ColorUtils.cpp
index 070e325..3812afe 100644
--- a/media/libstagefright/foundation/ColorUtils.cpp
+++ b/media/libstagefright/foundation/ColorUtils.cpp
@@ -613,6 +613,35 @@
}
// static
+void ColorUtils::getColorConfigFromDataSpace(
+ const android_dataspace &dataspace, int32_t *range, int32_t *standard, int32_t *transfer) {
+ uint32_t gfxRange =
+ (dataspace & HAL_DATASPACE_RANGE_MASK) >> HAL_DATASPACE_RANGE_SHIFT;
+ uint32_t gfxStandard =
+ (dataspace & HAL_DATASPACE_STANDARD_MASK) >> HAL_DATASPACE_STANDARD_SHIFT;
+ uint32_t gfxTransfer =
+ (dataspace & HAL_DATASPACE_TRANSFER_MASK) >> HAL_DATASPACE_TRANSFER_SHIFT;
+
+ // assume 1-to-1 mapping to HAL values (to deal with potential vendor extensions)
+ CU::ColorRange cuRange = CU::kColorRangeUnspecified;
+ CU::ColorStandard cuStandard = CU::kColorStandardUnspecified;
+ CU::ColorTransfer cuTransfer = CU::kColorTransferUnspecified;
+ // TRICKY: use & to ensure all three mappings are completed
+ if (!(sGfxRanges.map(gfxRange, &cuRange) & sGfxStandards.map(gfxStandard, &cuStandard)
+ & sGfxTransfers.map(gfxTransfer, &cuTransfer))) {
+ ALOGW("could not safely map graphics dataspace (R:%u S:%u T:%u) to "
+ "platform color aspects (R:%u(%s) S:%u(%s) T:%u(%s)",
+ gfxRange, gfxStandard, gfxTransfer,
+ cuRange, asString(cuRange),
+ cuStandard, asString(cuStandard),
+ cuTransfer, asString(cuTransfer));
+ }
+ *range = cuRange;
+ *standard = cuStandard;
+ *transfer = cuTransfer;
+}
+
+// static
void ColorUtils::getColorConfigFromFormat(
const sp<AMessage> &format, int32_t *range, int32_t *standard, int32_t *transfer) {
if (!format->findInt32("color-range", range)) {
diff --git a/media/libstagefright/foundation/include/media/stagefright/foundation/ColorUtils.h b/media/libstagefright/foundation/include/media/stagefright/foundation/ColorUtils.h
index cd0af2b..9e3f718 100644
--- a/media/libstagefright/foundation/include/media/stagefright/foundation/ColorUtils.h
+++ b/media/libstagefright/foundation/include/media/stagefright/foundation/ColorUtils.h
@@ -156,6 +156,10 @@
// suited to blending. This requires implicit color space conversion on part of the device.
static android_dataspace getDataSpaceForColorAspects(ColorAspects &aspects, bool mayExpand);
+ // it returns the platform color configs from given |dataspace|.
+ static void getColorConfigFromDataSpace(
+ const android_dataspace &dataspace, int *range, int *standard, int *transfer);
+
// converts |dataSpace| to a V0 enum, and returns true if dataSpace is an aspect-only value
static bool convertDataSpaceToV0(android_dataspace &dataSpace);
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index cd3c743..13e2ced 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -353,7 +353,8 @@
#endif
//ALOGD("Eric FastMixer::onWork() mIsWarm");
} else {
- dumpState->mTimestampVerifier.discontinuity();
+ dumpState->mTimestampVerifier.discontinuity(
+ dumpState->mTimestampVerifier.DISCONTINUITY_MODE_CONTINUOUS);
// See comment in if block.
#ifdef FASTMIXER_LOG_HIST_TS
LOG_AUDIO_STATE();
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 7804822..472c359 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -284,8 +284,6 @@
};
sp<AudioVibrationController> mAudioVibrationController;
sp<os::ExternalVibration> mExternalVibration;
- /** How many frames should be in the buffer before the track is considered ready */
- const size_t mFrameCountToBeReady;
audio_dual_mono_mode_t mDualMonoMode = AUDIO_DUAL_MONO_MODE_OFF;
float mAudioDescriptionMixLevel = -std::numeric_limits<float>::infinity();
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 927d87e..64e935d 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -622,7 +622,7 @@
mIoJitterMs.reset();
mLatencyMs.reset();
mProcessTimeMs.reset();
- mTimestampVerifier.discontinuity();
+ mTimestampVerifier.discontinuity(mTimestampVerifier.DISCONTINUITY_MODE_CONTINUOUS);
sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid, portId);
sendConfigEvent_l(configEvent);
@@ -2719,7 +2719,7 @@
// the timestamp frame position to reset to 0 for direct and offload threads.
// (Out of sequence requests are ignored, since the discontinuity would be handled
// elsewhere, e.g. in flush).
- mTimestampVerifier.discontinuity();
+ mTimestampVerifier.discontinuity(mTimestampVerifier.DISCONTINUITY_MODE_ZERO);
mDrainSequence &= ~1;
mWaitWorkCV.signal();
}
@@ -3408,7 +3408,6 @@
mStandbyTimeNs = systemTime();
int64_t lastLoopCountWritten = -2; // never matches "previous" loop, when loopCount = 0.
- int64_t lastFramesWritten = -1; // track changes in timestamp server frames written
// MIXER
nsecs_t lastWarning = 0;
@@ -3444,14 +3443,6 @@
checkSilentMode_l();
- // DIRECT and OFFLOAD threads should reset frame count to zero on stop/flush
- // TODO: add confirmation checks:
- // 1) DIRECT threads and linear PCM format really resets to 0?
- // 2) Is frame count really valid if not linear pcm?
- // 3) Are all 64 bits of position returned, not just lowest 32 bits?
- if (mType == OFFLOAD || mType == DIRECT) {
- mTimestampVerifier.setDiscontinuityMode(mTimestampVerifier.DISCONTINUITY_MODE_ZERO);
- }
audio_patch_handle_t lastDownstreamPatchHandle = AUDIO_PATCH_HANDLE_NONE;
// loopCount is used for statistics and diagnostics.
@@ -3523,135 +3514,8 @@
logString = NULL;
}
- // Collect timestamp statistics for the Playback Thread types that support it.
- if (mType == MIXER
- || mType == DUPLICATING
- || mType == DIRECT
- || mType == OFFLOAD) { // no indentation
- // Gather the framesReleased counters for all active tracks,
- // and associate with the sink frames written out. We need
- // this to convert the sink timestamp to the track timestamp.
- bool kernelLocationUpdate = false;
- ExtendedTimestamp timestamp; // use private copy to fetch
- if (mStandby) {
- mTimestampVerifier.discontinuity();
- } else if (threadloop_getHalTimestamp_l(×tamp) == OK) {
- mTimestampVerifier.add(timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL],
- timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
- mSampleRate);
+ collectTimestamps_l();
- if (isTimestampCorrectionEnabled()) {
- ALOGVV("TS_BEFORE: %d %lld %lld", id(),
- (long long)timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
- (long long)timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]);
- auto correctedTimestamp = mTimestampVerifier.getLastCorrectedTimestamp();
- timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
- = correctedTimestamp.mFrames;
- timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]
- = correctedTimestamp.mTimeNs;
- ALOGVV("TS_AFTER: %d %lld %lld", id(),
- (long long)timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
- (long long)timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]);
-
- // Note: Downstream latency only added if timestamp correction enabled.
- if (mDownstreamLatencyStatMs.getN() > 0) { // we have latency info.
- const int64_t newPosition =
- timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
- - int64_t(mDownstreamLatencyStatMs.getMean() * mSampleRate * 1e-3);
- // prevent retrograde
- timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = max(
- newPosition,
- (mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
- - mSuspendedFrames));
- }
- }
-
- // We always fetch the timestamp here because often the downstream
- // sink will block while writing.
-
- // We keep track of the last valid kernel position in case we are in underrun
- // and the normal mixer period is the same as the fast mixer period, or there
- // is some error from the HAL.
- if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
- mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
- mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
- mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
- mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
-
- mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
- mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER];
- mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
- mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER];
- }
-
- if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
- kernelLocationUpdate = true;
- } else {
- ALOGVV("getTimestamp error - no valid kernel position");
- }
-
- // copy over kernel info
- mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
- timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
- + mSuspendedFrames; // add frames discarded when suspended
- mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
- timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
- } else {
- mTimestampVerifier.error();
- }
-
- // mFramesWritten for non-offloaded tracks are contiguous
- // even after standby() is called. This is useful for the track frame
- // to sink frame mapping.
- bool serverLocationUpdate = false;
- if (mFramesWritten != lastFramesWritten) {
- serverLocationUpdate = true;
- lastFramesWritten = mFramesWritten;
- }
- // Only update timestamps if there is a meaningful change.
- // Either the kernel timestamp must be valid or we have written something.
- if (kernelLocationUpdate || serverLocationUpdate) {
- if (serverLocationUpdate) {
- // use the time before we called the HAL write - it is a bit more accurate
- // to when the server last read data than the current time here.
- //
- // If we haven't written anything, mLastIoBeginNs will be -1
- // and we use systemTime().
- mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten;
- mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastIoBeginNs == -1
- ? systemTime() : mLastIoBeginNs;
- }
-
- for (const sp<Track> &t : mActiveTracks) {
- if (!t->isFastTrack()) {
- t->updateTrackFrameInfo(
- t->mAudioTrackServerProxy->framesReleased(),
- mFramesWritten,
- mSampleRate,
- mTimestamp);
- }
- }
- }
-
- if (audio_has_proportional_frames(mFormat)) {
- const double latencyMs = mTimestamp.getOutputServerLatencyMs(mSampleRate);
- if (latencyMs != 0.) { // note 0. means timestamp is empty.
- mLatencyMs.add(latencyMs);
- }
- }
-
- } // if (mType ... ) { // no indentation
-#if 0
- // logFormat example
- if (z % 100 == 0) {
- timespec ts;
- clock_gettime(CLOCK_MONOTONIC, &ts);
- LOGT("This is an integer %d, this is a float %f, this is my "
- "pid %p %% %s %t", 42, 3.14, "and this is a timestamp", ts);
- LOGT("A deceptive null-terminated string %\0");
- }
- ++z;
-#endif
saveOutputTracks();
if (mSignalPending) {
// A signal was raised while we were unlocked
@@ -4091,6 +3955,148 @@
return false;
}
+void AudioFlinger::PlaybackThread::collectTimestamps_l()
+{
+ // Collect timestamp statistics for the Playback Thread types that support it.
+ if (mType != MIXER
+ && mType != DUPLICATING
+ && mType != DIRECT
+ && mType != OFFLOAD) {
+ return;
+ }
+ if (mStandby) {
+ mTimestampVerifier.discontinuity(discontinuityForStandbyOrFlush());
+ return;
+ } else if (mHwPaused) {
+ mTimestampVerifier.discontinuity(mTimestampVerifier.DISCONTINUITY_MODE_CONTINUOUS);
+ return;
+ }
+
+ // Gather the framesReleased counters for all active tracks,
+ // and associate with the sink frames written out. We need
+ // this to convert the sink timestamp to the track timestamp.
+ bool kernelLocationUpdate = false;
+ ExtendedTimestamp timestamp; // use private copy to fetch
+
+ // Always query HAL timestamp and update timestamp verifier. In standby or pause,
+ // HAL may be draining some small duration buffered data for fade out.
+ if (threadloop_getHalTimestamp_l(×tamp) == OK) {
+ mTimestampVerifier.add(timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL],
+ timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
+ mSampleRate);
+
+ if (isTimestampCorrectionEnabled()) {
+ ALOGVV("TS_BEFORE: %d %lld %lld", id(),
+ (long long)timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
+ (long long)timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]);
+ auto correctedTimestamp = mTimestampVerifier.getLastCorrectedTimestamp();
+ timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
+ = correctedTimestamp.mFrames;
+ timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]
+ = correctedTimestamp.mTimeNs;
+ ALOGVV("TS_AFTER: %d %lld %lld", id(),
+ (long long)timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
+ (long long)timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]);
+
+ // Note: Downstream latency only added if timestamp correction enabled.
+ if (mDownstreamLatencyStatMs.getN() > 0) { // we have latency info.
+ const int64_t newPosition =
+ timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
+ - int64_t(mDownstreamLatencyStatMs.getMean() * mSampleRate * 1e-3);
+ // prevent retrograde
+ timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = max(
+ newPosition,
+ (mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
+ - mSuspendedFrames));
+ }
+ }
+
+ // We always fetch the timestamp here because often the downstream
+ // sink will block while writing.
+
+ // We keep track of the last valid kernel position in case we are in underrun
+ // and the normal mixer period is the same as the fast mixer period, or there
+ // is some error from the HAL.
+ if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
+ mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
+ mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
+ mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
+ mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
+
+ mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
+ mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER];
+ mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
+ mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER];
+ }
+
+ if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
+ kernelLocationUpdate = true;
+ } else {
+ ALOGVV("getTimestamp error - no valid kernel position");
+ }
+
+ // copy over kernel info
+ mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
+ timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
+ + mSuspendedFrames; // add frames discarded when suspended
+ mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
+ timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
+ } else {
+ mTimestampVerifier.error();
+ }
+
+ // mFramesWritten for non-offloaded tracks are contiguous
+ // even after standby() is called. This is useful for the track frame
+ // to sink frame mapping.
+ bool serverLocationUpdate = false;
+ if (mFramesWritten != mLastFramesWritten) {
+ serverLocationUpdate = true;
+ mLastFramesWritten = mFramesWritten;
+ }
+ // Only update timestamps if there is a meaningful change.
+ // Either the kernel timestamp must be valid or we have written something.
+ if (kernelLocationUpdate || serverLocationUpdate) {
+ if (serverLocationUpdate) {
+ // use the time before we called the HAL write - it is a bit more accurate
+ // to when the server last read data than the current time here.
+ //
+ // If we haven't written anything, mLastIoBeginNs will be -1
+ // and we use systemTime().
+ mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten;
+ mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastIoBeginNs == -1
+ ? systemTime() : mLastIoBeginNs;
+ }
+
+ for (const sp<Track> &t : mActiveTracks) {
+ if (!t->isFastTrack()) {
+ t->updateTrackFrameInfo(
+ t->mAudioTrackServerProxy->framesReleased(),
+ mFramesWritten,
+ mSampleRate,
+ mTimestamp);
+ }
+ }
+ }
+
+ if (audio_has_proportional_frames(mFormat)) {
+ const double latencyMs = mTimestamp.getOutputServerLatencyMs(mSampleRate);
+ if (latencyMs != 0.) { // note 0. means timestamp is empty.
+ mLatencyMs.add(latencyMs);
+ }
+ }
+#if 0
+ // logFormat example
+ if (z % 100 == 0) {
+ timespec ts;
+ clock_gettime(CLOCK_MONOTONIC, &ts);
+ LOGT("This is an integer %d, this is a float %f, this is my "
+ "pid %p %% %s %t", 42, 3.14, "and this is a timestamp", ts);
+ LOGT("A deceptive null-terminated string %\0");
+ }
+ ++z;
+#endif
+}
+
// removeTracks_l() must be called with ThreadBase::mLock held
void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove)
{
@@ -4137,20 +4143,15 @@
return status;
}
if ((mType == OFFLOAD || mType == DIRECT) && mOutput != NULL) {
- uint64_t position64;
- if (mOutput->getPresentationPosition(&position64, ×tamp.mTime) == OK) {
- timestamp.mPosition = (uint32_t)position64;
- if (mDownstreamLatencyStatMs.getN() > 0) {
- const uint32_t positionOffset =
- (uint32_t)(mDownstreamLatencyStatMs.getMean() * mSampleRate * 1e-3);
- if (positionOffset > timestamp.mPosition) {
- timestamp.mPosition = 0;
- } else {
- timestamp.mPosition -= positionOffset;
- }
- }
- return NO_ERROR;
+ collectTimestamps_l();
+ if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] <= 0) {
+ return INVALID_OPERATION;
}
+ timestamp.mPosition = mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
+ const int64_t timeNs = mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
+ timestamp.mTime.tv_sec = timeNs / NANOS_PER_SECOND;
+ timestamp.mTime.tv_nsec = timeNs - (timestamp.mTime.tv_sec * NANOS_PER_SECOND);
+ return NO_ERROR;
}
return INVALID_OPERATION;
}
@@ -5551,8 +5552,6 @@
status_t& status)
{
bool reconfig = false;
- bool a2dpDeviceChanged = false;
-
status = NO_ERROR;
AutoPark<FastMixer> park(mFastMixer);
@@ -5624,7 +5623,7 @@
}
}
- return reconfig || a2dpDeviceChanged;
+ return reconfig;
}
@@ -6085,8 +6084,6 @@
status_t& status)
{
bool reconfig = false;
- bool a2dpDeviceChanged = false;
-
status = NO_ERROR;
AudioParameter param = AudioParameter(keyValuePair);
@@ -6121,7 +6118,7 @@
}
}
- return reconfig || a2dpDeviceChanged;
+ return reconfig;
}
uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
@@ -6178,7 +6175,7 @@
mOutput->flush();
mHwPaused = false;
mFlushPending = false;
- mTimestampVerifier.discontinuity(); // DIRECT and OFFLOADED flush resets frame count.
+ mTimestampVerifier.discontinuity(discontinuityForStandbyOrFlush());
mTimestamp.clear();
}
@@ -6514,13 +6511,14 @@
track->presentationComplete(framesWritten, audioHALFrames);
track->reset();
tracksToRemove->add(track);
- // DIRECT and OFFLOADED stop resets frame counts.
+ // OFFLOADED stop resets frame counts.
if (!mUseAsyncWrite) {
// If we don't get explicit drain notification we must
// register discontinuity regardless of whether this is
// the previous (!last) or the upcoming (last) track
// to avoid skipping the discontinuity.
- mTimestampVerifier.discontinuity();
+ mTimestampVerifier.discontinuity(
+ mTimestampVerifier.DISCONTINUITY_MODE_ZERO);
}
}
} else {
@@ -7378,7 +7376,9 @@
if (mPipeSource.get() == nullptr /* don't obtain for FastCapture, could block */) {
int64_t position, time;
if (mStandby) {
- mTimestampVerifier.discontinuity();
+ mTimestampVerifier.discontinuity(audio_is_linear_pcm(mFormat) ?
+ mTimestampVerifier.DISCONTINUITY_MODE_CONTINUOUS :
+ mTimestampVerifier.DISCONTINUITY_MODE_ZERO);
} else if (mSource->getCapturePosition(&position, &time) == NO_ERROR
&& time > mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 709a3cc..9f65562 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -590,6 +590,11 @@
ExtendedTimestamp mTimestamp;
TimestampVerifier< // For timestamp statistics.
int64_t /* frame count */, int64_t /* time ns */> mTimestampVerifier;
+ // DIRECT and OFFLOAD threads should reset frame count to zero on stop/flush
+ // TODO: add confirmation checks:
+ // 1) DIRECT threads and linear PCM format really resets to 0?
+ // 2) Is frame count really valid if not linear pcm?
+ // 3) Are all 64 bits of position returned, not just lowest 32 bits?
// Timestamp corrected device should be a single device.
audio_devices_t mTimestampCorrectedDevice = AUDIO_DEVICE_NONE;
@@ -1023,6 +1028,8 @@
int64_t mBytesWritten;
int64_t mFramesWritten; // not reset on standby
+ int64_t mLastFramesWritten = -1; // track changes in timestamp
+ // server frames written.
int64_t mSuspendedFrames; // not reset on standby
// mHapticChannelMask and mHapticChannelCount will only be valid when the thread support
@@ -1035,6 +1042,14 @@
// copy rather than the one in AudioFlinger. This optimization saves a lock.
bool mMasterMute;
void setMasterMute_l(bool muted) { mMasterMute = muted; }
+
+ auto discontinuityForStandbyOrFlush() const { // call on threadLoop or with lock.
+ return ((mType == DIRECT && !audio_is_linear_pcm(mFormat))
+ || mType == OFFLOAD)
+ ? mTimestampVerifier.DISCONTINUITY_MODE_ZERO
+ : mTimestampVerifier.DISCONTINUITY_MODE_CONTINUOUS;
+ }
+
protected:
ActiveTracks<Track> mActiveTracks;
@@ -1081,6 +1096,8 @@
void updateMetadata_l() final;
virtual void sendMetadataToBackend_l(const StreamOutHalInterface::SourceMetadata& metadata);
+ void collectTimestamps_l();
+
// The Tracks class manages tracks added and removed from the Thread.
template <typename T>
class Tracks {
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index ee886d5..fb43a6e 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -638,7 +638,6 @@
mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(
uid, attr, id(), streamType, opPackageName)),
// mSinkTimestamp
- mFrameCountToBeReady(frameCountToBeReady),
mFastIndex(-1),
mCachedVolume(1.0),
/* The track might not play immediately after being active, similarly as if its volume was 0.
@@ -672,6 +671,7 @@
mFrameSize, sampleRate);
}
mServerProxy = mAudioTrackServerProxy;
+ mServerProxy->setStartThresholdInFrames(frameCountToBeReady); // update the Cblk value
// only allocate a fast track index if we were able to allocate a normal track name
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
@@ -999,7 +999,10 @@
}
size_t bufferSizeInFrames = mServerProxy->getBufferSizeInFrames();
- size_t framesToBeReady = std::min(mFrameCountToBeReady, bufferSizeInFrames);
+ // Note: mServerProxy->getStartThresholdInFrames() is clamped.
+ const size_t startThresholdInFrames = mServerProxy->getStartThresholdInFrames();
+ const size_t framesToBeReady = std::clamp( // clamp again to validate client values.
+ std::min(startThresholdInFrames, bufferSizeInFrames), size_t(1), mFrameCount);
if (framesReady() >= framesToBeReady || (mCblk->mFlags & CBLK_FORCEREADY)) {
ALOGV("%s(%d): consider track ready with %zu/%zu, target was %zu)",
@@ -1038,6 +1041,11 @@
// initial state-stopping. next state-pausing.
// What if resume is called ?
+ if (state == FLUSHED) {
+ // avoid underrun glitches when starting after flush
+ reset();
+ }
+
if (state == PAUSED || state == PAUSING) {
if (mResumeToStopping) {
// happened we need to resume to STOPPING_1
diff --git a/services/mediametrics/Android.bp b/services/mediametrics/Android.bp
index b2a0cda..f13ca74 100644
--- a/services/mediametrics/Android.bp
+++ b/services/mediametrics/Android.bp
@@ -61,6 +61,7 @@
"-bugprone-unhandled-self-assignment", // found in TimeMachine.h
"-bugprone-suspicious-string-compare", // found in TimeMachine.h
"-cert-oop54-cpp", // found in TransactionLog.h
+ "-bugprone-narrowing-conversions", // b/182410845
]
cc_defaults {