audio: Provide a way for Module to specify nominal latency
The latency figure depends on the module implementation.
Instead of using a hardcoded value, each module should be
able to specify its own value. This value is then used
for calculating the minimum buffer size.
Set the nominal latency of the primary (CF) module to a high
value since the virtual device implementation fails CTS tests
if it attempts to pretend that it provides low latency.
Bug: 302132812
Test: atest CtsMediaAudioTestCases --test-filter=".*AudioTrackTest.*"
Test: atest CtsMediaAudioTestCases --test-filter=".*AudioRecordTest.*"
Change-Id: I8ce9f230378eea787c9b3c7ce3660c1e4e7bc895
diff --git a/audio/aidl/default/Module.cpp b/audio/aidl/default/Module.cpp
index a77397d..f725670 100644
--- a/audio/aidl/default/Module.cpp
+++ b/audio/aidl/default/Module.cpp
@@ -202,15 +202,17 @@
LOG(ERROR) << __func__ << ": non-positive buffer size " << in_bufferSizeFrames;
return ndk::ScopedAStatus::fromExceptionCode(EX_ILLEGAL_ARGUMENT);
}
- if (in_bufferSizeFrames < kMinimumStreamBufferSizeFrames) {
- LOG(ERROR) << __func__ << ": insufficient buffer size " << in_bufferSizeFrames
- << ", must be at least " << kMinimumStreamBufferSizeFrames;
- return ndk::ScopedAStatus::fromExceptionCode(EX_ILLEGAL_ARGUMENT);
- }
auto& configs = getConfig().portConfigs;
auto portConfigIt = findById<AudioPortConfig>(configs, in_portConfigId);
// Since this is a private method, it is assumed that
// validity of the portConfigId has already been checked.
+ const int32_t minimumStreamBufferSizeFrames = calculateBufferSizeFrames(
+ getNominalLatencyMs(*portConfigIt), portConfigIt->sampleRate.value().value);
+ if (in_bufferSizeFrames < minimumStreamBufferSizeFrames) {
+ LOG(ERROR) << __func__ << ": insufficient buffer size " << in_bufferSizeFrames
+ << ", must be at least " << minimumStreamBufferSizeFrames;
+ return ndk::ScopedAStatus::fromExceptionCode(EX_ILLEGAL_ARGUMENT);
+ }
const size_t frameSize =
getFrameSizeInBytes(portConfigIt->format.value(), portConfigIt->channelMask.value());
if (frameSize == 0) {
@@ -238,8 +240,8 @@
StreamContext temp(
std::make_unique<StreamContext::CommandMQ>(1, true /*configureEventFlagWord*/),
std::make_unique<StreamContext::ReplyMQ>(1, true /*configureEventFlagWord*/),
- portConfigIt->portId, portConfigIt->format.value(),
- portConfigIt->channelMask.value(), portConfigIt->sampleRate.value().value, flags,
+ portConfigIt->format.value(), portConfigIt->channelMask.value(),
+ portConfigIt->sampleRate.value().value, flags, getNominalLatencyMs(*portConfigIt),
portConfigIt->ext.get<AudioPortExt::mix>().handle,
std::make_unique<StreamContext::DataMQ>(frameSize * in_bufferSizeFrames),
asyncCallback, outEventCallback,
@@ -359,6 +361,12 @@
return internal::getConfiguration(getType());
}
+int32_t Module::getNominalLatencyMs(const AudioPortConfig&) {
+ // Arbitrary value. Implementations must override this method to provide their actual latency.
+ static constexpr int32_t kLatencyMs = 5;
+ return kLatencyMs;
+}
+
std::vector<AudioRoute*> Module::getAudioRoutesForAudioPortImpl(int32_t portId) {
std::vector<AudioRoute*> result;
auto& routes = getConfig().routes;
@@ -965,11 +973,21 @@
return ndk::ScopedAStatus::fromExceptionCode(EX_ILLEGAL_STATE);
}
}
+ // Find the highest sample rate among mix port configs.
+ std::map<int32_t, AudioPortConfig*> sampleRates;
+ std::vector<AudioPortConfig*>& mixPortConfigs =
+ sources[0]->ext.getTag() == AudioPortExt::mix ? sources : sinks;
+ for (auto mix : mixPortConfigs) {
+ sampleRates.emplace(mix->sampleRate.value().value, mix);
+ }
*_aidl_return = in_requested;
- _aidl_return->minimumStreamBufferSizeFrames = kMinimumStreamBufferSizeFrames;
+ auto maxSampleRateIt = std::max_element(sampleRates.begin(), sampleRates.end());
+ const int32_t latencyMs = getNominalLatencyMs(*(maxSampleRateIt->second));
+ _aidl_return->minimumStreamBufferSizeFrames =
+ calculateBufferSizeFrames(latencyMs, maxSampleRateIt->first);
_aidl_return->latenciesMs.clear();
_aidl_return->latenciesMs.insert(_aidl_return->latenciesMs.end(),
- _aidl_return->sinkPortConfigIds.size(), kLatencyMs);
+ _aidl_return->sinkPortConfigIds.size(), latencyMs);
AudioPatch oldPatch{};
if (existing == patches.end()) {
_aidl_return->id = getConfig().nextPatchId++;
diff --git a/audio/aidl/default/ModulePrimary.cpp b/audio/aidl/default/ModulePrimary.cpp
index 9919c7f..3da6d48 100644
--- a/audio/aidl/default/ModulePrimary.cpp
+++ b/audio/aidl/default/ModulePrimary.cpp
@@ -58,4 +58,12 @@
offloadInfo);
}
+int32_t ModulePrimary::getNominalLatencyMs(const AudioPortConfig&) {
+ // 85 ms is chosen considering 4096 frames @ 48 kHz. This is the value which allows
+ // the virtual Android device implementation to pass CTS. Hardware implementations
+ // should have significantly lower latency.
+ static constexpr int32_t kLatencyMs = 85;
+ return kLatencyMs;
+}
+
} // namespace aidl::android::hardware::audio::core
diff --git a/audio/aidl/default/Stream.cpp b/audio/aidl/default/Stream.cpp
index f00e358..6595324 100644
--- a/audio/aidl/default/Stream.cpp
+++ b/audio/aidl/default/Stream.cpp
@@ -138,7 +138,7 @@
reply->status = STATUS_OK;
if (isConnected) {
reply->observable.frames = mContext->getFrameCount();
- reply->observable.timeNs = ::android::elapsedRealtimeNano();
+ reply->observable.timeNs = ::android::uptimeNanos();
if (auto status = mDriver->refinePosition(&reply->observable); status == ::android::OK) {
return;
}
@@ -315,7 +315,7 @@
const size_t frameSize = mContext->getFrameSize();
size_t actualFrameCount = 0;
bool fatal = false;
- int32_t latency = Module::kLatencyMs;
+ int32_t latency = mContext->getNominalLatencyMs();
if (isConnected) {
if (::android::status_t status = mDriver->transfer(mDataBuffer.get(), byteCount / frameSize,
&actualFrameCount, &latency);
@@ -581,7 +581,7 @@
const size_t readByteCount = dataMQ->availableToRead();
const size_t frameSize = mContext->getFrameSize();
bool fatal = false;
- int32_t latency = Module::kLatencyMs;
+ int32_t latency = mContext->getNominalLatencyMs();
if (bool success = readByteCount > 0 ? dataMQ->read(&mDataBuffer[0], readByteCount) : true) {
const bool isConnected = mIsConnected;
LOG(VERBOSE) << __func__ << ": reading of " << readByteCount << " bytes from data MQ"
diff --git a/audio/aidl/default/alsa/StreamAlsa.cpp b/audio/aidl/default/alsa/StreamAlsa.cpp
index 403b94b..e57d538 100644
--- a/audio/aidl/default/alsa/StreamAlsa.cpp
+++ b/audio/aidl/default/alsa/StreamAlsa.cpp
@@ -119,7 +119,7 @@
::android::status_t StreamAlsa::refinePosition(StreamDescriptor::Position* position) {
if (mAlsaDeviceProxies.empty()) {
- LOG(FATAL) << __func__ << ": no opened devices";
+ LOG(WARNING) << __func__ << ": no opened devices";
return ::android::NO_INIT;
}
// Since the proxy can only count transferred frames since its creation,
diff --git a/audio/aidl/default/include/core-impl/Module.h b/audio/aidl/default/include/core-impl/Module.h
index d92d54b..8854f47 100644
--- a/audio/aidl/default/include/core-impl/Module.h
+++ b/audio/aidl/default/include/core-impl/Module.h
@@ -50,9 +50,6 @@
std::weak_ptr<IBluetoothLe>>
BtProfileHandles;
- // This value is used by default for all AudioPatches and reported by all streams.
- static constexpr int32_t kLatencyMs = 10;
-
static std::shared_ptr<Module> createInstance(Type type) {
return createInstance(type, std::make_unique<Configuration>());
}
@@ -145,8 +142,6 @@
ndk::ScopedAStatus getAAudioMixerBurstCount(int32_t* _aidl_return) override;
ndk::ScopedAStatus getAAudioHardwareBurstMinUsec(int32_t* _aidl_return) override;
- // This value is used for all AudioPatches.
- static constexpr int32_t kMinimumStreamBufferSizeFrames = 256;
// The maximum stream buffer size is 1 GiB = 2 ** 30 bytes;
static constexpr int32_t kMaximumStreamBufferSizeBytes = 1 << 30;
@@ -207,8 +202,17 @@
virtual ndk::ScopedAStatus onMasterVolumeChanged(float volume);
virtual std::vector<::aidl::android::media::audio::common::MicrophoneInfo> getMicrophoneInfos();
virtual std::unique_ptr<Configuration> initializeConfig();
+ virtual int32_t getNominalLatencyMs(
+ const ::aidl::android::media::audio::common::AudioPortConfig& portConfig);
// Utility and helper functions accessible to subclasses.
+ static int32_t calculateBufferSizeFrames(int32_t latencyMs, int32_t sampleRateHz) {
+ const int32_t rawSizeFrames = (latencyMs * sampleRateHz) / 1000;
+ int32_t powerOf2 = 1;
+ while (powerOf2 < rawSizeFrames) powerOf2 <<= 1;
+ return powerOf2;
+ }
+
ndk::ScopedAStatus bluetoothParametersUpdated();
void cleanUpPatch(int32_t patchId);
ndk::ScopedAStatus createStreamContext(
diff --git a/audio/aidl/default/include/core-impl/ModulePrimary.h b/audio/aidl/default/include/core-impl/ModulePrimary.h
index ee86d64..82c8a03 100644
--- a/audio/aidl/default/include/core-impl/ModulePrimary.h
+++ b/audio/aidl/default/include/core-impl/ModulePrimary.h
@@ -39,6 +39,8 @@
const std::optional<::aidl::android::media::audio::common::AudioOffloadInfo>&
offloadInfo,
std::shared_ptr<StreamOut>* result) override;
+ int32_t getNominalLatencyMs(
+ const ::aidl::android::media::audio::common::AudioPortConfig& portConfig) override;
private:
ChildInterface<ITelephony> mTelephony;
diff --git a/audio/aidl/default/include/core-impl/ModuleRemoteSubmix.h b/audio/aidl/default/include/core-impl/ModuleRemoteSubmix.h
index ebf4558..9f8acc9 100644
--- a/audio/aidl/default/include/core-impl/ModuleRemoteSubmix.h
+++ b/audio/aidl/default/include/core-impl/ModuleRemoteSubmix.h
@@ -50,6 +50,9 @@
override;
ndk::ScopedAStatus onMasterMuteChanged(bool mute) override;
ndk::ScopedAStatus onMasterVolumeChanged(float volume) override;
+ int32_t getNominalLatencyMs(
+ const ::aidl::android::media::audio::common::AudioPortConfig& portConfig) override;
+ // TODO(b/307586684): Report proper minimum stream buffer size by overriding 'setAudioPatch'.
};
} // namespace aidl::android::hardware::audio::core
diff --git a/audio/aidl/default/include/core-impl/Stream.h b/audio/aidl/default/include/core-impl/Stream.h
index daa920d..aa9fb19 100644
--- a/audio/aidl/default/include/core-impl/Stream.h
+++ b/audio/aidl/default/include/core-impl/Stream.h
@@ -81,11 +81,10 @@
StreamContext() = default;
StreamContext(std::unique_ptr<CommandMQ> commandMQ, std::unique_ptr<ReplyMQ> replyMQ,
- int portId,
const ::aidl::android::media::audio::common::AudioFormatDescription& format,
const ::aidl::android::media::audio::common::AudioChannelLayout& channelLayout,
int sampleRate, const ::aidl::android::media::audio::common::AudioIoFlags& flags,
- int32_t mixPortHandle, std::unique_ptr<DataMQ> dataMQ,
+ int32_t nominalLatencyMs, int32_t mixPortHandle, std::unique_ptr<DataMQ> dataMQ,
std::shared_ptr<IStreamCallback> asyncCallback,
std::shared_ptr<IStreamOutEventCallback> outEventCallback,
std::weak_ptr<sounddose::StreamDataProcessorInterface> streamDataProcessor,
@@ -93,51 +92,17 @@
: mCommandMQ(std::move(commandMQ)),
mInternalCommandCookie(std::rand()),
mReplyMQ(std::move(replyMQ)),
- mPortId(portId),
mFormat(format),
mChannelLayout(channelLayout),
mSampleRate(sampleRate),
mFlags(flags),
+ mNominalLatencyMs(nominalLatencyMs),
mMixPortHandle(mixPortHandle),
mDataMQ(std::move(dataMQ)),
mAsyncCallback(asyncCallback),
mOutEventCallback(outEventCallback),
mStreamDataProcessor(streamDataProcessor),
mDebugParameters(debugParameters) {}
- StreamContext(StreamContext&& other)
- : mCommandMQ(std::move(other.mCommandMQ)),
- mInternalCommandCookie(other.mInternalCommandCookie),
- mReplyMQ(std::move(other.mReplyMQ)),
- mPortId(other.mPortId),
- mFormat(other.mFormat),
- mChannelLayout(other.mChannelLayout),
- mSampleRate(other.mSampleRate),
- mFlags(std::move(other.mFlags)),
- mMixPortHandle(other.mMixPortHandle),
- mDataMQ(std::move(other.mDataMQ)),
- mAsyncCallback(std::move(other.mAsyncCallback)),
- mOutEventCallback(std::move(other.mOutEventCallback)),
- mStreamDataProcessor(std::move(other.mStreamDataProcessor)),
- mDebugParameters(std::move(other.mDebugParameters)),
- mFrameCount(other.mFrameCount) {}
- StreamContext& operator=(StreamContext&& other) {
- mCommandMQ = std::move(other.mCommandMQ);
- mInternalCommandCookie = other.mInternalCommandCookie;
- mReplyMQ = std::move(other.mReplyMQ);
- mPortId = std::move(other.mPortId);
- mFormat = std::move(other.mFormat);
- mChannelLayout = std::move(other.mChannelLayout);
- mSampleRate = other.mSampleRate;
- mFlags = std::move(other.mFlags);
- mMixPortHandle = other.mMixPortHandle;
- mDataMQ = std::move(other.mDataMQ);
- mAsyncCallback = std::move(other.mAsyncCallback);
- mOutEventCallback = std::move(other.mOutEventCallback);
- mStreamDataProcessor = std::move(other.mStreamDataProcessor);
- mDebugParameters = std::move(other.mDebugParameters);
- mFrameCount = other.mFrameCount;
- return *this;
- }
void fillDescriptor(StreamDescriptor* desc);
std::shared_ptr<IStreamCallback> getAsyncCallback() const { return mAsyncCallback; }
@@ -156,6 +121,7 @@
size_t getFrameSize() const;
int getInternalCommandCookie() const { return mInternalCommandCookie; }
int32_t getMixPortHandle() const { return mMixPortHandle; }
+ int32_t getNominalLatencyMs() const { return mNominalLatencyMs; }
std::shared_ptr<IStreamOutEventCallback> getOutEventCallback() const {
return mOutEventCallback;
}
@@ -163,7 +129,6 @@
return mStreamDataProcessor;
}
void startStreamDataProcessor();
- int getPortId() const { return mPortId; }
ReplyMQ* getReplyMQ() const { return mReplyMQ.get(); }
int getTransientStateDelayMs() const { return mDebugParameters.transientStateDelayMs; }
int getSampleRate() const { return mSampleRate; }
@@ -176,14 +141,15 @@
long getFrameCount() const { return mFrameCount; }
private:
+ // Fields are non const to allow move assignment.
std::unique_ptr<CommandMQ> mCommandMQ;
int mInternalCommandCookie; // The value used to confirm that the command was posted internally
std::unique_ptr<ReplyMQ> mReplyMQ;
- int mPortId;
::aidl::android::media::audio::common::AudioFormatDescription mFormat;
::aidl::android::media::audio::common::AudioChannelLayout mChannelLayout;
int mSampleRate;
::aidl::android::media::audio::common::AudioIoFlags mFlags;
+ int32_t mNominalLatencyMs;
int32_t mMixPortHandle;
std::unique_ptr<DataMQ> mDataMQ;
std::shared_ptr<IStreamCallback> mAsyncCallback;
diff --git a/audio/aidl/default/include/core-impl/StreamPrimary.h b/audio/aidl/default/include/core-impl/StreamPrimary.h
index b64b749..abc119c 100644
--- a/audio/aidl/default/include/core-impl/StreamPrimary.h
+++ b/audio/aidl/default/include/core-impl/StreamPrimary.h
@@ -27,10 +27,13 @@
public:
StreamPrimary(StreamContext* context, const Metadata& metadata);
+ ::android::status_t transfer(void* buffer, size_t frameCount, size_t* actualFrameCount,
+ int32_t* latencyMs) override;
+
protected:
std::vector<alsa::DeviceProfile> getDeviceProfiles() override;
- const bool mIsInput;
+ const bool mIsAsynchronous;
};
class StreamInPrimary final : public StreamIn, public StreamSwitcher, public StreamInHwGainHelper {
diff --git a/audio/aidl/default/primary/StreamPrimary.cpp b/audio/aidl/default/primary/StreamPrimary.cpp
index 17de2ba..1b1ea68 100644
--- a/audio/aidl/default/primary/StreamPrimary.cpp
+++ b/audio/aidl/default/primary/StreamPrimary.cpp
@@ -14,11 +14,12 @@
* limitations under the License.
*/
-#include <limits>
+#include <chrono>
#define LOG_TAG "AHAL_StreamPrimary"
#include <android-base/logging.h>
#include <android-base/properties.h>
+#include <audio_utils/clock.h>
#include <error/expected_utils.h>
#include "PrimaryMixer.h"
@@ -37,10 +38,34 @@
namespace aidl::android::hardware::audio::core {
StreamPrimary::StreamPrimary(StreamContext* context, const Metadata& metadata)
- : StreamAlsa(context, metadata, 3 /*readWriteRetries*/), mIsInput(isInput(metadata)) {
+ : StreamAlsa(context, metadata, 3 /*readWriteRetries*/),
+ mIsAsynchronous(!!getContext().getAsyncCallback()) {
context->startStreamDataProcessor();
}
+::android::status_t StreamPrimary::transfer(void* buffer, size_t frameCount,
+ size_t* actualFrameCount, int32_t* latencyMs) {
+ auto start = std::chrono::steady_clock::now();
+ if (auto status = StreamAlsa::transfer(buffer, frameCount, actualFrameCount, latencyMs);
+ status != ::android::OK) {
+ return status;
+ }
+ // This is a workaround for the emulator implementation which has a host-side buffer
+ // and this can result in reading faster than real time.
+ if (mIsInput && !mIsAsynchronous) {
+ auto recordDurationUs = std::chrono::duration_cast<std::chrono::microseconds>(
+ std::chrono::steady_clock::now() - start);
+ const long projectedVsObservedOffsetUs =
+ *actualFrameCount * MICROS_PER_SECOND / mContext.getSampleRate() -
+ recordDurationUs.count();
+ if (projectedVsObservedOffsetUs > 0) {
+ LOG(VERBOSE) << __func__ << ": sleeping for " << projectedVsObservedOffsetUs << " us";
+ usleep(projectedVsObservedOffsetUs);
+ }
+ }
+ return ::android::OK;
+}
+
std::vector<alsa::DeviceProfile> StreamPrimary::getDeviceProfiles() {
static const std::vector<alsa::DeviceProfile> kBuiltInSource{
alsa::DeviceProfile{.card = primary::PrimaryMixer::kAlsaCard,
@@ -66,7 +91,8 @@
GetBoolProperty("ro.boot.audio.tinyalsa.simulate_input", false);
return kSimulateInput || device.type.type == AudioDeviceType::IN_TELEPHONY_RX ||
device.type.type == AudioDeviceType::IN_FM_TUNER ||
- device.type.connection == AudioDeviceDescription::CONNECTION_BUS;
+ device.type.connection == AudioDeviceDescription::CONNECTION_BUS ||
+ (device.type.type == AudioDeviceType::IN_DEVICE && device.type.connection.empty());
}
StreamSwitcher::DeviceSwitchBehavior StreamInPrimary::switchCurrentStream(
@@ -132,7 +158,8 @@
static const bool kSimulateOutput =
GetBoolProperty("ro.boot.audio.tinyalsa.ignore_output", false);
return kSimulateOutput || device.type.type == AudioDeviceType::OUT_TELEPHONY_TX ||
- device.type.connection == AudioDeviceDescription::CONNECTION_BUS;
+ device.type.connection == AudioDeviceDescription::CONNECTION_BUS ||
+ (device.type.type == AudioDeviceType::OUT_DEVICE && device.type.connection.empty());
}
StreamSwitcher::DeviceSwitchBehavior StreamOutPrimary::switchCurrentStream(
diff --git a/audio/aidl/default/r_submix/ModuleRemoteSubmix.cpp b/audio/aidl/default/r_submix/ModuleRemoteSubmix.cpp
index f8c775f..3e8dd7c 100644
--- a/audio/aidl/default/r_submix/ModuleRemoteSubmix.cpp
+++ b/audio/aidl/default/r_submix/ModuleRemoteSubmix.cpp
@@ -21,6 +21,7 @@
#include <android-base/logging.h>
#include <error/expected_utils.h>
+#include "SubmixRoute.h"
#include "core-impl/ModuleRemoteSubmix.h"
#include "core-impl/StreamRemoteSubmix.h"
@@ -106,4 +107,12 @@
return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
}
+int32_t ModuleRemoteSubmix::getNominalLatencyMs(const AudioPortConfig&) {
+ // See the note on kDefaultPipePeriodCount.
+ static constexpr int32_t kMaxLatencyMs =
+ (r_submix::kDefaultPipeSizeInFrames * 1000) / r_submix::kDefaultSampleRateHz;
+ static constexpr int32_t kMinLatencyMs = kMaxLatencyMs / r_submix::kDefaultPipePeriodCount;
+ return (kMaxLatencyMs + kMinLatencyMs) / 2;
+}
+
} // namespace aidl::android::hardware::audio::core
diff --git a/audio/aidl/default/r_submix/SubmixRoute.h b/audio/aidl/default/r_submix/SubmixRoute.h
index 1fe9ea2..92b95e9 100644
--- a/audio/aidl/default/r_submix/SubmixRoute.h
+++ b/audio/aidl/default/r_submix/SubmixRoute.h
@@ -39,9 +39,13 @@
namespace aidl::android::hardware::audio::core::r_submix {
static constexpr int kDefaultSampleRateHz = 48000;
-// Size at default sample rate
-// NOTE: This value will be rounded up to the nearest power of 2 by MonoPipe().
-static constexpr int kDefaultPipeSizeInFrames = (1024 * 4);
+// Value used to divide the MonoPipe buffer into segments that are written to the source and
+// read from the sink. The maximum latency of the device is the size of the MonoPipe's buffer
+// the minimum latency is the MonoPipe buffer size divided by this value.
+static constexpr int kDefaultPipePeriodCount = 4;
+// Size at the default sample rate
+// NOTE: This value will be rounded up to the nearest power of 2 by MonoPipe.
+static constexpr int kDefaultPipeSizeInFrames = 1024 * kDefaultPipePeriodCount;
// Configuration of the audio stream.
struct AudioConfig {
diff --git a/audio/aidl/default/stub/StreamStub.cpp b/audio/aidl/default/stub/StreamStub.cpp
index 660a51e..2422fe4 100644
--- a/audio/aidl/default/stub/StreamStub.cpp
+++ b/audio/aidl/default/stub/StreamStub.cpp
@@ -94,7 +94,7 @@
}
::android::status_t StreamStub::transfer(void* buffer, size_t frameCount, size_t* actualFrameCount,
- int32_t* latencyMs) {
+ int32_t*) {
if (!mIsInitialized) {
LOG(FATAL) << __func__ << ": must not happen for an uninitialized driver";
}
@@ -117,7 +117,6 @@
}
}
*actualFrameCount = frameCount;
- *latencyMs = Module::kLatencyMs;
return ::android::OK;
}