Merge "Pause faster during audio underrun" into tm-qpr-dev
diff --git a/camera/ndk/include/camera/NdkCameraMetadataTags.h b/camera/ndk/include/camera/NdkCameraMetadataTags.h
index b6f8552..0d156a5 100644
--- a/camera/ndk/include/camera/NdkCameraMetadataTags.h
+++ b/camera/ndk/include/camera/NdkCameraMetadataTags.h
@@ -2198,6 +2198,10 @@
* <a href="https://developer.android.com/reference/android/hardware/camera2/CameraManager.html#turnOnTorchWithStrengthLevel">CameraManager#turnOnTorchWithStrengthLevel</a>.
* If this value is equal to 1, flashlight brightness control is not supported.
* The value for this key will be null for devices with no flash unit.</p>
+ * <p>The maximum value is guaranteed to be safe to use for an indefinite duration in
+ * terms of device flashlight lifespan, but may be too bright for comfort for many
+ * use cases. Use the default torch brightness value to avoid problems with an
+ * over-bright flashlight.</p>
*/
ACAMERA_FLASH_INFO_STRENGTH_MAXIMUM_LEVEL = // int32
ACAMERA_FLASH_INFO_START + 2,
diff --git a/media/TEST_MAPPING b/media/TEST_MAPPING
index a4c03ba..048301f 100644
--- a/media/TEST_MAPPING
+++ b/media/TEST_MAPPING
@@ -23,8 +23,51 @@
{
"path": "frameworks/av/drm/mediadrm/plugins"
}
+ ],
+
+ "platinum-postsubmit": [
+ // runs regularly, independent of changes in this tree.
+ // signals if changes elsewhere break media functionality
+ // @FlakyTest: in staged-postsubmit, but not postsubmit
+ {
+ "name": "CtsMediaCodecTestCases",
+ "options": [
+ {
+ "include-filter": "android.media.codec.cts.EncodeDecodeTest"
+ }
+ ]
+ },
+ {
+ "name": "CtsMediaCodecTestCases",
+ "options": [
+ {
+ "include-filter": "android.media.codec.cts.DecodeEditEncodeTest"
+ },
+ {
+ "exclude-annotation": "androidx.test.filters.FlakyTest"
+ }
+ ]
+ }
+ ],
+
+ "staged-platinum-postsubmit": [
+ // runs every four hours
+ {
+ "name": "CtsMediaCodecTestCases",
+ "options": [
+ {
+ "include-filter": "android.media.codec.cts.EncodeDecodeTest"
+ }
+ ]
+ },
+ {
+ "name": "CtsMediaCodecTestCases",
+ "options": [
+ {
+ "include-filter": "android.media.codec.cts.DecodeEditEncodeTest"
+ }
+ ]
+ }
]
- // TODO (b/229286407) Add EncodeDecodeTest and DecodeEditEncodeTest to
- // platinum-postsubmit once issues in cuttlefish are fixed
}
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index dc9f848..5ecb130 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -2041,6 +2041,9 @@
sp<MediaCodecBuffer> outBuffer;
std::shared_ptr<C2Buffer> c2Buffer;
+ constexpr int kMaxReallocTry = 5;
+ int reallocTryNum = 0;
+
while (true) {
Mutexed<Output>::Locked output(mOutput);
if (!output->buffers) {
@@ -2048,6 +2051,9 @@
}
action = output->buffers->popFromStashAndRegister(
&c2Buffer, &index, &outBuffer);
+ if (action != OutputBuffers::REALLOCATE) {
+ reallocTryNum = 0;
+ }
switch (action) {
case OutputBuffers::SKIP:
return;
@@ -2058,6 +2064,13 @@
mCallback->onOutputBufferAvailable(index, outBuffer);
break;
case OutputBuffers::REALLOCATE:
+ if (++reallocTryNum > kMaxReallocTry) {
+ output.unlock();
+ ALOGE("[%s] sendOutputBuffers: tried %d realloc and failed",
+ mName, kMaxReallocTry);
+ mCCodecCallback->onError(UNKNOWN_ERROR, ACTION_CODE_FATAL);
+ return;
+ }
if (!output->buffers->isArrayMode()) {
output->buffers =
output->buffers->toArrayMode(output->numSlots);
diff --git a/media/codec2/sfplugin/Codec2InfoBuilder.cpp b/media/codec2/sfplugin/Codec2InfoBuilder.cpp
index 1c362ae..453a0d2 100644
--- a/media/codec2/sfplugin/Codec2InfoBuilder.cpp
+++ b/media/codec2/sfplugin/Codec2InfoBuilder.cpp
@@ -156,9 +156,10 @@
// dynamic metadata as that needs to be frame accurate.)
supportsHdr |= (mediaType == MIMETYPE_VIDEO_VP9);
- // HDR support implies 10-bit support.
+ // HDR support implies 10-bit support. AV1 codecs are also required to
+ // support 10-bit per CDD.
// TODO: directly check this from the component interface
- supports10Bit = (supportsHdr || supportsHdr10Plus);
+ supports10Bit = (supportsHdr || supportsHdr10Plus) || (mediaType == MIMETYPE_VIDEO_AV1);
// If the device doesn't support HDR display, then no codec on the device
// can advertise support for HDR profiles.
diff --git a/media/libaudioclient/AidlConversion.cpp b/media/libaudioclient/AidlConversion.cpp
index 11724e0..4133bd0 100644
--- a/media/libaudioclient/AidlConversion.cpp
+++ b/media/libaudioclient/AidlConversion.cpp
@@ -3388,4 +3388,24 @@
enumToMask_index<int32_t, media::AudioDirectMode>);
}
+ConversionResult<audio_latency_mode_t>
+aidl2legacy_LatencyMode_audio_latency_mode_t(media::LatencyMode aidl) {
+ switch (aidl) {
+ case media::LatencyMode::FREE:
+ return AUDIO_LATENCY_MODE_FREE;
+ case media::LatencyMode::LOW:
+ return AUDIO_LATENCY_MODE_LOW;
+ }
+ return unexpected(BAD_VALUE);
+}
+ConversionResult<media::LatencyMode>
+legacy2aidl_audio_latency_mode_t_LatencyMode(audio_latency_mode_t legacy) {
+ switch (legacy) {
+ case AUDIO_LATENCY_MODE_FREE:
+ return media::LatencyMode::FREE;
+ case AUDIO_LATENCY_MODE_LOW:
+ return media::LatencyMode::LOW;
+ }
+ return unexpected(BAD_VALUE);
+}
} // namespace android
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 69a9c68..dccef0e 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -342,6 +342,7 @@
"aidl/android/media/AudioUniqueIdUse.aidl",
"aidl/android/media/AudioVibratorInfo.aidl",
"aidl/android/media/EffectDescriptor.aidl",
+ "aidl/android/media/LatencyMode.aidl",
"aidl/android/media/TrackSecondaryOutputInfo.aidl",
],
imports: [
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index de8c298..bacca3a 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -670,6 +670,30 @@
return Status::ok();
}
+Status AudioSystem::AudioFlingerClient::onSupportedLatencyModesChanged(
+ int output, const std::vector<media::LatencyMode>& latencyModes) {
+ audio_io_handle_t outputLegacy = VALUE_OR_RETURN_BINDER_STATUS(
+ aidl2legacy_int32_t_audio_io_handle_t(output));
+ std::vector<audio_latency_mode_t> modesLegacy = VALUE_OR_RETURN_BINDER_STATUS(
+ convertContainer<std::vector<audio_latency_mode_t>>(
+ latencyModes, aidl2legacy_LatencyMode_audio_latency_mode_t));
+
+ std::vector<sp<SupportedLatencyModesCallback>> callbacks;
+ {
+ Mutex::Autolock _l(mLock);
+ for (auto callback : mSupportedLatencyModesCallbacks) {
+ if (auto ref = callback.promote(); ref != nullptr) {
+ callbacks.push_back(ref);
+ }
+ }
+ }
+ for (const auto& callback : callbacks) {
+ callback->onSupportedLatencyModesChanged(outputLegacy, modesLegacy);
+ }
+
+ return Status::ok();
+}
+
status_t AudioSystem::AudioFlingerClient::getInputBufferSize(
uint32_t sampleRate, audio_format_t format,
audio_channel_mask_t channelMask, size_t* buffSize) {
@@ -749,6 +773,31 @@
return NO_ERROR;
}
+status_t AudioSystem::AudioFlingerClient::addSupportedLatencyModesCallback(
+ const sp<SupportedLatencyModesCallback>& callback) {
+ Mutex::Autolock _l(mLock);
+ if (std::find(mSupportedLatencyModesCallbacks.begin(),
+ mSupportedLatencyModesCallbacks.end(),
+ callback) != mSupportedLatencyModesCallbacks.end()) {
+ return INVALID_OPERATION;
+ }
+ mSupportedLatencyModesCallbacks.push_back(callback);
+ return NO_ERROR;
+}
+
+status_t AudioSystem::AudioFlingerClient::removeSupportedLatencyModesCallback(
+ const sp<SupportedLatencyModesCallback>& callback) {
+ Mutex::Autolock _l(mLock);
+ auto it = std::find(mSupportedLatencyModesCallbacks.begin(),
+ mSupportedLatencyModesCallbacks.end(),
+ callback);
+ if (it == mSupportedLatencyModesCallbacks.end()) {
+ return INVALID_OPERATION;
+ }
+ mSupportedLatencyModesCallbacks.erase(it);
+ return NO_ERROR;
+}
+
/* static */ uintptr_t AudioSystem::addErrorCallback(audio_error_callback cb) {
Mutex::Autolock _l(gLockErrorCallbacks);
gAudioErrorCallbacks.insert(cb);
@@ -1662,6 +1711,24 @@
return afc->removeAudioDeviceCallback(callback, audioIo, portId);
}
+status_t AudioSystem::addSupportedLatencyModesCallback(
+ const sp<SupportedLatencyModesCallback>& callback) {
+ const sp<AudioFlingerClient> afc = getAudioFlingerClient();
+ if (afc == 0) {
+ return NO_INIT;
+ }
+ return afc->addSupportedLatencyModesCallback(callback);
+}
+
+status_t AudioSystem::removeSupportedLatencyModesCallback(
+ const sp<SupportedLatencyModesCallback>& callback) {
+ const sp<AudioFlingerClient> afc = getAudioFlingerClient();
+ if (afc == 0) {
+ return NO_INIT;
+ }
+ return afc->removeSupportedLatencyModesCallback(callback);
+}
+
audio_port_handle_t AudioSystem::getDeviceIdForIo(audio_io_handle_t audioIo) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -2338,6 +2405,24 @@
return NO_ERROR;
}
+status_t AudioSystem::setRequestedLatencyMode(
+ audio_io_handle_t output, audio_latency_mode_t mode) {
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == nullptr) {
+ return PERMISSION_DENIED;
+ }
+ return af->setRequestedLatencyMode(output, mode);
+}
+
+status_t AudioSystem::getSupportedLatencyModes(audio_io_handle_t output,
+ std::vector<audio_latency_mode_t>* modes) {
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == nullptr) {
+ return PERMISSION_DENIED;
+ }
+ return af->getSupportedLatencyModes(output, modes);
+}
+
class CaptureStateListenerImpl : public media::BnCaptureStateListener,
public IBinder::DeathRecipient {
public:
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 292d92f..6ad97d1 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -810,6 +810,33 @@
return statusTFromBinderStatus(mDelegate->setDeviceConnectedState(aidlPort, connected));
}
+status_t AudioFlingerClientAdapter::setRequestedLatencyMode(
+ audio_io_handle_t output, audio_latency_mode_t mode) {
+ int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
+ media::LatencyMode modeAidl = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_latency_mode_t_LatencyMode(mode));
+ return statusTFromBinderStatus(mDelegate->setRequestedLatencyMode(outputAidl, modeAidl));
+}
+
+status_t AudioFlingerClientAdapter::getSupportedLatencyModes(
+ audio_io_handle_t output, std::vector<audio_latency_mode_t>* modes) {
+ if (modes == nullptr) {
+ return BAD_VALUE;
+ }
+
+ int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
+ std::vector<media::LatencyMode> modesAidl;
+
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->getSupportedLatencyModes(outputAidl, &modesAidl)));
+
+ *modes = VALUE_OR_RETURN_STATUS(
+ convertContainer<std::vector<audio_latency_mode_t>>(modesAidl,
+ aidl2legacy_LatencyMode_audio_latency_mode_t));
+
+ return NO_ERROR;
+}
+
////////////////////////////////////////////////////////////////////////////////////////////////////
// AudioFlingerServerAdapter
AudioFlingerServerAdapter::AudioFlingerServerAdapter(
@@ -1304,4 +1331,28 @@
return Status::fromStatusT(mDelegate->setDeviceConnectedState(&portLegacy, connected));
}
+Status AudioFlingerServerAdapter::setRequestedLatencyMode(
+ int32_t output, media::LatencyMode modeAidl) {
+ audio_io_handle_t outputLegacy = VALUE_OR_RETURN_BINDER(
+ aidl2legacy_int32_t_audio_io_handle_t(output));
+ audio_latency_mode_t modeLegacy = VALUE_OR_RETURN_BINDER(
+ aidl2legacy_LatencyMode_audio_latency_mode_t(modeAidl));
+ return Status::fromStatusT(mDelegate->setRequestedLatencyMode(
+ outputLegacy, modeLegacy));
+}
+
+Status AudioFlingerServerAdapter::getSupportedLatencyModes(
+ int output, std::vector<media::LatencyMode>* _aidl_return) {
+ audio_io_handle_t outputLegacy = VALUE_OR_RETURN_BINDER(
+ aidl2legacy_int32_t_audio_io_handle_t(output));
+ std::vector<audio_latency_mode_t> modesLegacy;
+
+ RETURN_BINDER_IF_ERROR(mDelegate->getSupportedLatencyModes(outputLegacy, &modesLegacy));
+
+ *_aidl_return = VALUE_OR_RETURN_BINDER(
+ convertContainer<std::vector<media::LatencyMode>>(
+ modesLegacy, legacy2aidl_audio_latency_mode_t_LatencyMode));
+ return Status::ok();
+}
+
} // namespace android
diff --git a/media/libaudioclient/aidl/android/media/IAudioFlingerClient.aidl b/media/libaudioclient/aidl/android/media/IAudioFlingerClient.aidl
index 421c31c..a2bb024 100644
--- a/media/libaudioclient/aidl/android/media/IAudioFlingerClient.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioFlingerClient.aidl
@@ -18,6 +18,7 @@
import android.media.AudioIoConfigEvent;
import android.media.AudioIoDescriptor;
+import android.media.LatencyMode;
/**
* A callback interface for AudioFlinger.
@@ -27,4 +28,10 @@
interface IAudioFlingerClient {
oneway void ioConfigChanged(AudioIoConfigEvent event,
in AudioIoDescriptor ioDesc);
+ /**
+ * Called when the latency modes supported on a given output stream change.
+ * output is the I/O handle of the output stream for which the change is signalled.
+ * latencyModes is the new list of supported latency modes (See LatencyMode.aidl).
+ */
+ oneway void onSupportedLatencyModesChanged(int output, in LatencyMode[] latencyModes);
}
diff --git a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
index 10da028..9b8a843 100644
--- a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
@@ -35,6 +35,7 @@
import android.media.IAudioFlingerClient;
import android.media.IAudioRecord;
import android.media.IAudioTrack;
+import android.media.LatencyMode;
import android.media.MicrophoneInfoData;
import android.media.RenderPosition;
import android.media.TrackSecondaryOutputInfo;
@@ -228,6 +229,23 @@
void setDeviceConnectedState(in AudioPort devicePort, boolean connected);
+ /**
+ * Requests a given latency mode (See LatencyMode.aidl) on an output stream.
+ * This can be used when some use case on a given mixer/stream can only be enabled
+ * if a specific latency mode is selected on the audio path below the HAL.
+ * For instance spatial audio with head tracking.
+ * output is the I/O handle of the output stream for which the request is made.
+ * latencyMode is the requested latency mode.
+ */
+ void setRequestedLatencyMode(int output, LatencyMode latencyMode);
+
+ /**
+ * Queries the list of latency modes (See LatencyMode.aidl) supported by an output stream.
+ * output is the I/O handle of the output stream to which the query applies.
+ * returns the list of supported latency modes.
+ */
+ LatencyMode[] getSupportedLatencyModes(int output);
+
// When adding a new method, please review and update
// IAudioFlinger.h AudioFlingerServerAdapter::Delegate::TransactionCode
// AudioFlinger.cpp AudioFlinger::onTransactWrapper()
diff --git a/media/libaudioclient/aidl/android/media/LatencyMode.aidl b/media/libaudioclient/aidl/android/media/LatencyMode.aidl
new file mode 100644
index 0000000..0b2a72b
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/LatencyMode.aidl
@@ -0,0 +1,29 @@
+/*
+ * Copyright 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+/**
+ * The latency mode currently used by the spatializer mixer.
+ * {@hide}
+ */
+@Backing(type="byte")
+enum LatencyMode {
+ /** No specific constraint on the latency */
+ FREE = 0,
+ /** A relatively low latency compatible with head tracking operation (e.g less than 100ms) */
+ LOW = 1,
+}
diff --git a/media/libaudioclient/include/media/AidlConversion.h b/media/libaudioclient/include/media/AidlConversion.h
index e769303..1e66164 100644
--- a/media/libaudioclient/include/media/AidlConversion.h
+++ b/media/libaudioclient/include/media/AidlConversion.h
@@ -35,6 +35,7 @@
#include <android/media/AudioTimestampInternal.h>
#include <android/media/AudioUniqueIdUse.h>
#include <android/media/EffectDescriptor.h>
+#include <android/media/LatencyMode.h>
#include <android/media/TrackSecondaryOutputInfo.h>
#include <android/media/audio/common/AudioChannelLayout.h>
#include <android/media/audio/common/AudioConfig.h>
@@ -454,5 +455,9 @@
ConversionResult<audio_direct_mode_t> aidl2legacy_int32_t_audio_direct_mode_t_mask(int32_t aidl);
ConversionResult<int32_t> legacy2aidl_audio_direct_mode_t_int32_t_mask(audio_direct_mode_t legacy);
+ConversionResult<audio_latency_mode_t>
+aidl2legacy_LatencyMode_audio_latency_mode_t(media::LatencyMode aidl);
+ConversionResult<media::LatencyMode>
+legacy2aidl_audio_latency_mode_t_LatencyMode(audio_latency_mode_t legacy);
} // namespace android
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 360b83d..9411f46 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -569,6 +569,12 @@
static status_t getDirectProfilesForAttributes(const audio_attributes_t* attr,
std::vector<audio_profile>* audioProfiles);
+ static status_t setRequestedLatencyMode(
+ audio_io_handle_t output, audio_latency_mode_t mode);
+
+ static status_t getSupportedLatencyModes(audio_io_handle_t output,
+ std::vector<audio_latency_mode_t>* modes);
+
// A listener for capture state changes.
class CaptureStateListener : public virtual RefBase {
public:
@@ -644,6 +650,22 @@
audio_io_handle_t audioIo,
audio_port_handle_t portId);
+ class SupportedLatencyModesCallback : public virtual RefBase
+ {
+ public:
+
+ SupportedLatencyModesCallback() = default;
+ virtual ~SupportedLatencyModesCallback() = default;
+
+ virtual void onSupportedLatencyModesChanged(
+ audio_io_handle_t output, const std::vector<audio_latency_mode_t>& modes) = 0;
+ };
+
+ static status_t addSupportedLatencyModesCallback(
+ const sp<SupportedLatencyModesCallback>& callback);
+ static status_t removeSupportedLatencyModesCallback(
+ const sp<SupportedLatencyModesCallback>& callback);
+
static audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo);
static status_t setVibratorInfos(const std::vector<media::AudioVibratorInfo>& vibratorInfos);
@@ -682,6 +704,9 @@
media::AudioIoConfigEvent event,
const media::AudioIoDescriptor& ioDesc) override;
+ binder::Status onSupportedLatencyModesChanged(
+ int output, const std::vector<media::LatencyMode>& latencyModes) override;
+
status_t addAudioDeviceCallback(const wp<AudioDeviceCallback>& callback,
audio_io_handle_t audioIo,
audio_port_handle_t portId);
@@ -689,6 +714,11 @@
audio_io_handle_t audioIo,
audio_port_handle_t portId);
+ status_t addSupportedLatencyModesCallback(
+ const sp<SupportedLatencyModesCallback>& callback);
+ status_t removeSupportedLatencyModesCallback(
+ const sp<SupportedLatencyModesCallback>& callback);
+
audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo);
private:
@@ -697,6 +727,10 @@
std::map<audio_io_handle_t, std::map<audio_port_handle_t, wp<AudioDeviceCallback>>>
mAudioDeviceCallbacks;
+
+ std::vector<wp<SupportedLatencyModesCallback>>
+ mSupportedLatencyModesCallbacks GUARDED_BY(mLock);
+
// cached values for recording getInputBufferSize() queries
size_t mInBuffSize; // zero indicates cache is invalid
uint32_t mInSamplingRate;
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 3c3715d..c891ae6 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -360,6 +360,13 @@
virtual int32_t getAAudioHardwareBurstMinUsec() = 0;
virtual status_t setDeviceConnectedState(const struct audio_port_v7 *port, bool connected) = 0;
+
+ virtual status_t setRequestedLatencyMode(
+ audio_io_handle_t output, audio_latency_mode_t mode) = 0;
+
+ virtual status_t getSupportedLatencyModes(audio_io_handle_t output,
+ std::vector<audio_latency_mode_t>* modes) = 0;
+
};
/**
@@ -462,6 +469,10 @@
int32_t getAAudioMixerBurstCount() override;
int32_t getAAudioHardwareBurstMinUsec() override;
status_t setDeviceConnectedState(const struct audio_port_v7 *port, bool connected) override;
+ status_t setRequestedLatencyMode(audio_io_handle_t output,
+ audio_latency_mode_t mode) override;
+ status_t getSupportedLatencyModes(
+ audio_io_handle_t output, std::vector<audio_latency_mode_t>* modes) override;
private:
const sp<media::IAudioFlingerService> mDelegate;
@@ -551,6 +562,8 @@
GET_AAUDIO_MIXER_BURST_COUNT = media::BnAudioFlingerService::TRANSACTION_getAAudioMixerBurstCount,
GET_AAUDIO_HARDWARE_BURST_MIN_USEC = media::BnAudioFlingerService::TRANSACTION_getAAudioHardwareBurstMinUsec,
SET_DEVICE_CONNECTED_STATE = media::BnAudioFlingerService::TRANSACTION_setDeviceConnectedState,
+ SET_REQUESTED_LATENCY_MODE = media::BnAudioFlingerService::TRANSACTION_setRequestedLatencyMode,
+ GET_SUPPORTED_LATENCY_MODES = media::BnAudioFlingerService::TRANSACTION_getSupportedLatencyModes,
};
protected:
@@ -672,7 +685,9 @@
Status getAAudioMixerBurstCount(int32_t* _aidl_return) override;
Status getAAudioHardwareBurstMinUsec(int32_t* _aidl_return) override;
Status setDeviceConnectedState(const media::AudioPort& port, bool connected) override;
-
+ Status setRequestedLatencyMode(int output, media::LatencyMode mode) override;
+ Status getSupportedLatencyModes(int output,
+ std::vector<media::LatencyMode>* _aidl_return) override;
private:
const sp<AudioFlingerServerAdapter::Delegate> mDelegate;
};
diff --git a/media/libaudiohal/impl/EffectHalHidl.cpp b/media/libaudiohal/impl/EffectHalHidl.cpp
index fdfe225..8743c04 100644
--- a/media/libaudiohal/impl/EffectHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectHalHidl.cpp
@@ -93,13 +93,13 @@
}
status_t EffectHalHidl::process() {
- TIME_CHECK();
+ // TIME_CHECK(); // TODO(b/238654698) reenable only when optimized.
return processImpl(static_cast<uint32_t>(MessageQueueFlagBits::REQUEST_PROCESS));
}
status_t EffectHalHidl::processReverse() {
- TIME_CHECK();
+ // TIME_CHECK(); // TODO(b/238654698) reenable only when optimized.
return processImpl(static_cast<uint32_t>(MessageQueueFlagBits::REQUEST_PROCESS_REVERSE));
}
diff --git a/media/libaudiohal/impl/StreamHalHidl.cpp b/media/libaudiohal/impl/StreamHalHidl.cpp
index 021ec51..76f9a60 100644
--- a/media/libaudiohal/impl/StreamHalHidl.cpp
+++ b/media/libaudiohal/impl/StreamHalHidl.cpp
@@ -441,7 +441,7 @@
#endif
status_t StreamOutHalHidl::write(const void *buffer, size_t bytes, size_t *written) {
- TIME_CHECK();
+ // TIME_CHECK(); // TODO(b/238654698) reenable only when optimized.
if (mStream == 0) return NO_INIT;
*written = 0;
@@ -587,7 +587,7 @@
}
status_t StreamOutHalHidl::getRenderPosition(uint32_t *dspFrames) {
- TIME_CHECK();
+ // TIME_CHECK(); // TODO(b/238654698) reenable only when optimized.
if (mStream == 0) return NO_INIT;
Result retval;
Return<void> ret = mStream->getRenderPosition(
@@ -668,7 +668,7 @@
}
status_t StreamOutHalHidl::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) {
- TIME_CHECK();
+ // TIME_CHECK(); // TODO(b/238654698) reenable only when optimized.
if (mStream == 0) return NO_INIT;
if (mWriterClient == gettid() && mCommandMQ) {
return callWriterThread(
@@ -1012,7 +1012,7 @@
}
status_t StreamInHalHidl::read(void *buffer, size_t bytes, size_t *read) {
- TIME_CHECK();
+ // TIME_CHECK(); // TODO(b/238654698) reenable only when optimized.
if (mStream == 0) return NO_INIT;
*read = 0;
@@ -1146,7 +1146,7 @@
}
status_t StreamInHalHidl::getCapturePosition(int64_t *frames, int64_t *time) {
- TIME_CHECK();
+ // TIME_CHECK(); // TODO(b/238654698) reenable only when optimized.
if (mStream == 0) return NO_INIT;
if (mReaderClient == gettid() && mCommandMQ) {
ReadParameters params;
diff --git a/media/libmediaplayerservice/nuplayer/AWakeLock.cpp b/media/libmediaplayerservice/nuplayer/AWakeLock.cpp
index c3bd207..25a8ae4 100644
--- a/media/libmediaplayerservice/nuplayer/AWakeLock.cpp
+++ b/media/libmediaplayerservice/nuplayer/AWakeLock.cpp
@@ -59,11 +59,10 @@
if (mPowerManager != NULL) {
sp<IBinder> binder = new BBinder();
int64_t token = IPCThreadState::self()->clearCallingIdentity();
- binder::Status status = mPowerManager->acquireWakeLock(
+ binder::Status status = mPowerManager->acquireWakeLockAsync(
binder, POWERMANAGER_PARTIAL_WAKE_LOCK,
String16("AWakeLock"), String16("media"),
- {} /* workSource */, {} /* historyTag */, -1 /* displayId */,
- nullptr /* callback */);
+ {} /* workSource */, {} /* historyTag */);
IPCThreadState::self()->restoreCallingIdentity(token);
if (status.isOk()) {
mWakeLockToken = binder;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 2828d44..45fac76 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -73,6 +73,10 @@
// is closed to allow the audio DSP to power down.
static const int64_t kOffloadPauseMaxUs = 10000000LL;
+// Additional delay after teardown before releasing the wake lock to allow time for the audio path
+// to be completely released
+static const int64_t kWakelockReleaseDelayUs = 2000000LL;
+
// Maximum allowed delay from AudioSink, 1.5 seconds.
static const int64_t kMaxAllowedAudioSinkDelayUs = 1500000LL;
@@ -793,6 +797,20 @@
}
ALOGV("Audio Offload tear down due to pause timeout.");
onAudioTearDown(kDueToTimeout);
+ sp<AMessage> newMsg = new AMessage(kWhatReleaseWakeLock, this);
+ newMsg->setInt32("drainGeneration", generation);
+ newMsg->post(kWakelockReleaseDelayUs);
+ break;
+ }
+
+ case kWhatReleaseWakeLock:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("drainGeneration", &generation));
+ if (generation != mAudioOffloadPauseTimeoutGeneration) {
+ break;
+ }
+ ALOGV("releasing audio offload pause wakelock.");
mWakeLock->release();
break;
}
@@ -1785,6 +1803,8 @@
return;
}
+ startAudioOffloadPauseTimeout();
+
{
Mutex::Autolock autoLock(mLock);
// we do not increment audio drain generation so that we fill audio buffer during pause.
@@ -1799,7 +1819,6 @@
// Note: audio data may not have been decoded, and the AudioSink may not be opened.
mAudioSink->pause();
- startAudioOffloadPauseTimeout();
ALOGV("now paused audio queue has %zu entries, video has %zu entries",
mAudioQueue.size(), mVideoQueue.size());
@@ -1927,12 +1946,27 @@
int32_t numChannels;
CHECK(format->findInt32("channel-count", &numChannels));
- int32_t rawChannelMask;
- audio_channel_mask_t channelMask =
- format->findInt32("channel-mask", &rawChannelMask) ?
- static_cast<audio_channel_mask_t>(rawChannelMask)
- // signal to the AudioSink to derive the mask from count.
- : CHANNEL_MASK_USE_CHANNEL_ORDER;
+ // channel mask info as read from the audio format
+ int32_t channelMaskFromFormat;
+ // channel mask to use for native playback
+ audio_channel_mask_t channelMask;
+ if (format->findInt32("channel-mask", &channelMaskFromFormat)) {
+ // KEY_CHANNEL_MASK follows the android.media.AudioFormat java mask
+ // which is left-bitshifted by 2 relative to the native mask
+ if ((channelMaskFromFormat & 0b11) != 0) {
+ // received an unexpected mask (supposed to follow AudioFormat constants
+ // for output masks with the 2 least-significant bits at 0), but
+ // it may come from an extractor that uses native masks: keeping
+ // the mask as given is ok as it contains at least mono or stereo
+ // and potentially the haptic channels
+ channelMask = static_cast<audio_channel_mask_t>(channelMaskFromFormat);
+ } else {
+ channelMask = static_cast<audio_channel_mask_t>(channelMaskFromFormat >> 2);
+ }
+ } else {
+ // no mask found: the mask will be derived from the channel count
+ channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
+ }
int32_t sampleRate;
CHECK(format->findInt32("sample-rate", &sampleRate));
diff --git a/media/libmediaplayerservice/nuplayer/include/nuplayer/NuPlayerRenderer.h b/media/libmediaplayerservice/nuplayer/include/nuplayer/NuPlayerRenderer.h
index 3d2b033..3640678 100644
--- a/media/libmediaplayerservice/nuplayer/include/nuplayer/NuPlayerRenderer.h
+++ b/media/libmediaplayerservice/nuplayer/include/nuplayer/NuPlayerRenderer.h
@@ -100,6 +100,7 @@
kWhatMediaRenderingStart = 'mdrd',
kWhatAudioTearDown = 'adTD',
kWhatAudioOffloadPauseTimeout = 'aOPT',
+ kWhatReleaseWakeLock = 'adRL',
};
enum AudioTearDownReason {
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index e50880a..c963e19 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -3534,6 +3534,17 @@
}
setState(STARTED);
postPendingRepliesAndDeferredMessages("kWhatStartCompleted");
+
+ // Now that the codec has started, configure, by default, the peek behavior to
+ // be undefined for backwards compatibility with older releases. Later, if an
+ // app explicitly enables or disables peek, the parameter will be turned off and
+ // the legacy undefined behavior is disallowed.
+ // See updateTunnelPeek called in onSetParameters for more details.
+ if (mTunneled && mTunnelPeekState == TunnelPeekState::kLegacyMode) {
+ sp<AMessage> params = new AMessage;
+ params->setInt32("android._tunnel-peek-set-legacy", 1);
+ mCodec->signalSetParameters(params);
+ }
break;
}
@@ -3973,14 +3984,6 @@
mTunneled = false;
}
- // If mTunnelPeekState is still in kLegacyMode at this point,
- // configure the codec in legacy mode
- if (mTunneled && (mTunnelPeekState == TunnelPeekState::kLegacyMode)) {
- sp<AMessage> params = new AMessage;
- params->setInt32("android._tunnel-peek-set-legacy", 1);
- onSetParameters(params);
- }
-
int32_t background = 0;
if (format->findInt32("android._background-mode", &background) && background) {
androidSetThreadPriority(gettid(), ANDROID_PRIORITY_BACKGROUND);
diff --git a/media/libstagefright/renderfright/gl/ProgramCache.cpp b/media/libstagefright/renderfright/gl/ProgramCache.cpp
index 3ae35ec..56d35a9 100644
--- a/media/libstagefright/renderfright/gl/ProgramCache.cpp
+++ b/media/libstagefright/renderfright/gl/ProgramCache.cpp
@@ -299,8 +299,8 @@
highp vec3 ScaleLuminance(highp vec3 color) {
// The formula is:
// alpha * pow(Y, gamma - 1.0) * color + beta;
- // where alpha is 1000.0, gamma is 1.2, beta is 0.0.
- return color * 1000.0 * pow(color.y, 0.2);
+ // where alpha is displayMaxLuminance, gamma is 1.2, beta is 0.0.
+ return color * displayMaxLuminance * pow(color.y, 0.2);
}
)__SHADER__";
break;
@@ -316,7 +316,6 @@
// Tone map absolute light to display luminance range.
switch (needs.getInputTF()) {
case Key::INPUT_TF_ST2084:
- case Key::INPUT_TF_HLG:
switch (needs.getOutputTF()) {
case Key::OUTPUT_TF_HLG:
// Right now when mixed PQ and HLG contents are presented,
@@ -396,6 +395,14 @@
break;
}
break;
+ case Key::INPUT_TF_HLG:
+ // HLG OOTF is already applied as part of ScaleLuminance
+ fs << R"__SHADER__(
+ highp vec3 ToneMap(highp vec3 color) {
+ return color;
+ }
+ )__SHADER__";
+ break;
default:
// inverse tone map; the output luminance can be up to maxOutLumi.
fs << R"__SHADER__(
diff --git a/media/libstagefright/timedtext/TextDescriptions.cpp b/media/libstagefright/timedtext/TextDescriptions.cpp
index 2c2d11d..3fec9ed 100644
--- a/media/libstagefright/timedtext/TextDescriptions.cpp
+++ b/media/libstagefright/timedtext/TextDescriptions.cpp
@@ -466,6 +466,10 @@
if (subChunkType == FOURCC('f', 't', 'a', 'b'))
{
+ if(subChunkSize < 8) {
+ return OK;
+ }
+
tmpData += 8;
size_t subChunkRemaining = subChunkSize - 8;
diff --git a/media/ndk/NdkMediaCodec.cpp b/media/ndk/NdkMediaCodec.cpp
index 38e422d..7495082 100644
--- a/media/ndk/NdkMediaCodec.cpp
+++ b/media/ndk/NdkMediaCodec.cpp
@@ -529,22 +529,31 @@
AMediaCodecOnAsyncNotifyCallback callback,
void *userdata) {
- Mutex::Autolock _l(mData->mAsyncCallbackLock);
-
- if (mData->mAsyncNotify == NULL) {
- mData->mAsyncNotify = new AMessage(kWhatAsyncNotify, mData->mHandler);
+ {
+ Mutex::Autolock _l(mData->mAsyncCallbackLock);
+ if (mData->mAsyncNotify == NULL) {
+ mData->mAsyncNotify = new AMessage(kWhatAsyncNotify, mData->mHandler);
+ }
+ // we set this ahead so that we can be ready
+ // to receive callbacks as soon as the next call is a
+ // success.
+ mData->mAsyncCallback = callback;
+ mData->mAsyncCallbackUserData = userdata;
}
// always call, codec may have been reset/re-configured since last call.
status_t err = mData->mCodec->setCallback(mData->mAsyncNotify);
if (err != OK) {
+ {
+ //The setup gone wrong. clean up the pointers.
+ Mutex::Autolock _l(mData->mAsyncCallbackLock);
+ mData->mAsyncCallback = {};
+ mData->mAsyncCallbackUserData = nullptr;
+ }
ALOGE("setAsyncNotifyCallback: err(%d), failed to set async callback", err);
return translate_error(err);
}
- mData->mAsyncCallback = callback;
- mData->mAsyncCallbackUserData = userdata;
-
return AMEDIA_OK;
}
diff --git a/media/ndk/NdkMediaDrm.cpp b/media/ndk/NdkMediaDrm.cpp
index 59c1103..f4674de 100644
--- a/media/ndk/NdkMediaDrm.cpp
+++ b/media/ndk/NdkMediaDrm.cpp
@@ -183,7 +183,7 @@
AMediaDrmSessionId asid = {sessionId.data(), sessionId.size()};
int32_t dataSize = data.size();
const uint8_t *dataPtr = data.data();
- if (dataSize > 0) {
+ if (dataSize >= 0) {
(*mEventListener)(mObj, &asid, ndkEventType, 0, dataPtr, dataSize);
} else {
ALOGE("invalid event data size=%d", dataSize);
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index f7576f6..8ae06d0 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -226,6 +226,9 @@
BINDER_METHOD_ENTRY(getAAudioMixerBurstCount) \
BINDER_METHOD_ENTRY(getAAudioHardwareBurstMinUsec) \
BINDER_METHOD_ENTRY(setDeviceConnectedState) \
+BINDER_METHOD_ENTRY(setRequestedLatencyMode) \
+BINDER_METHOD_ENTRY(getSupportedLatencyModes) \
+
// singleton for Binder Method Statistics for IAudioFlinger
static auto& getIAudioFlingerStatistics() {
@@ -1583,6 +1586,32 @@
return NO_ERROR;
}
+status_t AudioFlinger::setRequestedLatencyMode(
+ audio_io_handle_t output, audio_latency_mode_t mode) {
+ if (output == AUDIO_IO_HANDLE_NONE) {
+ return BAD_VALUE;
+ }
+ AutoMutex lock(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == nullptr) {
+ return BAD_VALUE;
+ }
+ return thread->setRequestedLatencyMode(mode);
+}
+
+status_t AudioFlinger::getSupportedLatencyModes(audio_io_handle_t output,
+ std::vector<audio_latency_mode_t>* modes) {
+ if (output == AUDIO_IO_HANDLE_NONE) {
+ return BAD_VALUE;
+ }
+ AutoMutex lock(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == nullptr) {
+ return BAD_VALUE;
+ }
+ return thread->getSupportedLatencyModes(modes);
+}
+
status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
{
// check calling permissions
@@ -2082,6 +2111,21 @@
}
}
+void AudioFlinger::onSupportedLatencyModesChanged(
+ audio_io_handle_t output, const std::vector<audio_latency_mode_t>& modes) {
+ int32_t outputAidl = VALUE_OR_FATAL(legacy2aidl_audio_io_handle_t_int32_t(output));
+ std::vector<media::LatencyMode> modesAidl = VALUE_OR_FATAL(
+ convertContainer<std::vector<media::LatencyMode>>(modes,
+ legacy2aidl_audio_latency_mode_t_LatencyMode));
+
+ Mutex::Autolock _l(mClientLock);
+ size_t size = mNotificationClients.size();
+ for (size_t i = 0; i < size; i++) {
+ mNotificationClients.valueAt(i)->audioFlingerClient()
+ ->onSupportedLatencyModesChanged(outputAidl, modesAidl);
+ }
+}
+
// removeClient_l() must be called with AudioFlinger::mClientLock held
void AudioFlinger::removeClient_l(pid_t pid)
{
@@ -4476,6 +4520,8 @@
case TransactionCode::SET_RECORD_SILENCED:
case TransactionCode::AUDIO_POLICY_READY:
case TransactionCode::SET_DEVICE_CONNECTED_STATE:
+ case TransactionCode::SET_REQUESTED_LATENCY_MODE:
+ case TransactionCode::GET_SUPPORTED_LATENCY_MODES:
ALOGW("%s: transaction %d received from PID %d",
__func__, code, IPCThreadState::self()->getCallingPid());
// return status only for non void methods
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 8e4383c..fc4c807 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -293,6 +293,12 @@
virtual status_t setDeviceConnectedState(const struct audio_port_v7 *port, bool connected);
+ virtual status_t setRequestedLatencyMode(
+ audio_io_handle_t output, audio_latency_mode_t mode);
+
+ virtual status_t getSupportedLatencyModes(audio_io_handle_t output,
+ std::vector<audio_latency_mode_t>* modes);
+
status_t onTransactWrapper(TransactionCode code, const Parcel& data, uint32_t flags,
const std::function<status_t()>& delegate) override;
@@ -764,6 +770,8 @@
void ioConfigChanged(audio_io_config_event_t event,
const sp<AudioIoDescriptor>& ioDesc,
pid_t pid = 0);
+ void onSupportedLatencyModesChanged(
+ audio_io_handle_t output, const std::vector<audio_latency_mode_t>& modes);
// Allocate an audio_unique_id_t.
// Specific types are audio_io_handle_t, audio_session_t, effect ID (int),
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 401a19e..56ee014 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -746,6 +746,12 @@
sendConfigEvent_l(configEvent);
}
+void AudioFlinger::ThreadBase::sendHalLatencyModesChangedEvent_l()
+{
+ sp<ConfigEvent> configEvent = sp<HalLatencyModesChangedEvent>::make();
+ sendConfigEvent_l(configEvent);
+}
+
// post condition: mConfigEvents.isEmpty()
void AudioFlinger::ThreadBase::processConfigEvents_l()
{
@@ -813,6 +819,10 @@
setCheckOutputStageEffects();
} break;
+ case CFG_EVENT_HAL_LATENCY_MODES_CHANGED: {
+ onHalLatencyModesChanged_l();
+ } break;
+
default:
ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
break;
@@ -3142,10 +3152,7 @@
auto backInserter = std::back_inserter(metadata.tracks);
for (const sp<Track> &track : mActiveTracks) {
// No track is invalid as this is called after prepareTrack_l in the same critical section
- // Do not forward metadata for PatchTrack with unspecified stream type
- if (track->streamType() != AUDIO_STREAM_PATCH) {
- track->copyMetadataTo(backInserter);
- }
+ track->copyMetadataTo(backInserter);
}
sendMetadataToBackend_l(metadata);
}
@@ -3928,6 +3935,8 @@
// stop(), pause(), etc.), but the threadLoop is entitled to call audio
// data / buffer methods on tracks from activeTracks without the ThreadBase lock.
activeTracks.insert(activeTracks.end(), mActiveTracks.begin(), mActiveTracks.end());
+
+ setHalLatencyMode_l();
} // mLock scope ends
if (mBytesRemaining == 0) {
@@ -3973,19 +3982,24 @@
void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
- // mono blend occurs for mixer threads only (not direct or offloaded)
- // and is handled here if we're going directly to the sink.
- if (requireMonoBlend() && !mEffectBufferValid) {
- mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount,
- true /*limit*/);
- }
+ // Apply mono blending and balancing if the effect buffer is not valid. Otherwise,
+ // do these processes after effects are applied.
+ if (!mEffectBufferValid) {
+ // mono blend occurs for mixer threads only (not direct or offloaded)
+ // and is handled here if we're going directly to the sink.
+ if (requireMonoBlend()) {
+ mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount,
+ mNormalFrameCount, true /*limit*/);
+ }
- if (!hasFastMixer()) {
- // Balance must take effect after mono conversion.
- // We do it here if there is no FastMixer.
- // mBalance detects zero balance within the class for speed (not needed here).
- mBalance.setBalance(mMasterBalance.load());
- mBalance.process((float *)mMixerBuffer, mNormalFrameCount);
+ if (!hasFastMixer()) {
+ // Balance must take effect after mono conversion.
+ // We do it here if there is no FastMixer.
+ // mBalance detects zero balance within the class for speed
+ // (not needed here).
+ mBalance.setBalance(mMasterBalance.load());
+ mBalance.process((float *)mMixerBuffer, mNormalFrameCount);
+ }
}
memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
@@ -5022,6 +5036,7 @@
mCallbackThread->setDraining(mDrainSequence);
}
mHwPaused = false;
+ setHalLatencyMode_l();
}
void AudioFlinger::PlaybackThread::onAddNewTrack_l()
@@ -7282,6 +7297,94 @@
{
}
+void AudioFlinger::SpatializerThread::onFirstRef() {
+ PlaybackThread::onFirstRef();
+
+ Mutex::Autolock _l(mLock);
+ status_t status = mOutput->stream->setLatencyModeCallback(this);
+ if (status != INVALID_OPERATION) {
+ updateHalSupportedLatencyModes_l();
+ }
+}
+
+status_t AudioFlinger::SpatializerThread::createAudioPatch_l(const struct audio_patch *patch,
+ audio_patch_handle_t *handle)
+{
+ status_t status = MixerThread::createAudioPatch_l(patch, handle);
+ updateHalSupportedLatencyModes_l();
+ return status;
+}
+
+void AudioFlinger::SpatializerThread::updateHalSupportedLatencyModes_l() {
+ std::vector<audio_latency_mode_t> latencyModes;
+ if (mOutput->stream->getRecommendedLatencyModes(&latencyModes) != NO_ERROR) {
+ latencyModes.clear();
+ }
+ if (latencyModes != mSupportedLatencyModes) {
+ mSupportedLatencyModes.swap(latencyModes);
+ sendHalLatencyModesChangedEvent_l();
+ }
+}
+
+void AudioFlinger::SpatializerThread::onHalLatencyModesChanged_l() {
+ mAudioFlinger->onSupportedLatencyModesChanged(mId, mSupportedLatencyModes);
+}
+
+void AudioFlinger::SpatializerThread::setHalLatencyMode_l() {
+ // if mSupportedLatencyModes is empty, the HAL stream does not support
+ // latency mode control and we can exit.
+ if (mSupportedLatencyModes.empty()) {
+ return;
+ }
+ audio_latency_mode_t latencyMode = AUDIO_LATENCY_MODE_FREE;
+ if (mSupportedLatencyModes.size() == 1) {
+ // If the HAL only support one latency mode currently, confirm the choice
+ latencyMode = mSupportedLatencyModes[0];
+ } else if (mSupportedLatencyModes.size() > 1) {
+ // Request low latency if:
+ // - The low latency mode is requested by the spatializer controller
+ // (mRequestedLatencyMode = AUDIO_LATENCY_MODE_LOW)
+ // AND
+ // - At least one active track is spatialized
+ bool hasSpatializedActiveTrack = false;
+ for (const auto& track : mActiveTracks) {
+ if (track->isSpatialized()) {
+ hasSpatializedActiveTrack = true;
+ break;
+ }
+ }
+ if (hasSpatializedActiveTrack && mRequestedLatencyMode == AUDIO_LATENCY_MODE_LOW) {
+ latencyMode = AUDIO_LATENCY_MODE_LOW;
+ }
+ }
+
+ if (latencyMode != mSetLatencyMode) {
+ status_t status = mOutput->stream->setLatencyMode(latencyMode);
+ if (status == NO_ERROR) {
+ mSetLatencyMode = latencyMode;
+ }
+ }
+}
+
+status_t AudioFlinger::SpatializerThread::setRequestedLatencyMode(audio_latency_mode_t mode) {
+ if (mode != AUDIO_LATENCY_MODE_LOW && mode != AUDIO_LATENCY_MODE_FREE) {
+ return BAD_VALUE;
+ }
+ Mutex::Autolock _l(mLock);
+ mRequestedLatencyMode = mode;
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::SpatializerThread::getSupportedLatencyModes(
+ std::vector<audio_latency_mode_t>* modes) {
+ if (modes == nullptr) {
+ return BAD_VALUE;
+ }
+ Mutex::Autolock _l(mLock);
+ *modes = mSupportedLatencyModes;
+ return NO_ERROR;
+}
+
void AudioFlinger::SpatializerThread::checkOutputStageEffects()
{
bool hasVirtualizer = false;
@@ -7334,6 +7437,14 @@
}
}
+void AudioFlinger::SpatializerThread::onRecommendedLatencyModeChanged(
+ std::vector<audio_latency_mode_t> modes) {
+ Mutex::Autolock _l(mLock);
+ if (modes != mSupportedLatencyModes) {
+ mSupportedLatencyModes.swap(modes);
+ sendHalLatencyModesChangedEvent_l();
+ }
+}
// ----------------------------------------------------------------------------
// Record
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index e196e1e..da5adea 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -55,7 +55,8 @@
CFG_EVENT_RELEASE_AUDIO_PATCH,
CFG_EVENT_UPDATE_OUT_DEVICE,
CFG_EVENT_RESIZE_BUFFER,
- CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS
+ CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS,
+ CFG_EVENT_HAL_LATENCY_MODES_CHANGED,
};
class ConfigEventData: public RefBase {
@@ -282,6 +283,15 @@
virtual ~CheckOutputStageEffectsEvent() {}
};
+ class HalLatencyModesChangedEvent : public ConfigEvent {
+ public:
+ HalLatencyModesChangedEvent() :
+ ConfigEvent(CFG_EVENT_HAL_LATENCY_MODES_CHANGED) {
+ }
+
+ virtual ~HalLatencyModesChangedEvent() {}
+ };
+
class PMDeathRecipient : public IBinder::DeathRecipient {
public:
@@ -353,6 +363,7 @@
void sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs);
void sendCheckOutputStageEffectsEvent();
void sendCheckOutputStageEffectsEvent_l();
+ void sendHalLatencyModesChangedEvent_l();
void processConfigEvents_l();
virtual void setCheckOutputStageEffects() {}
@@ -364,7 +375,7 @@
virtual void toAudioPortConfig(struct audio_port_config *config) = 0;
virtual void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs);
-
+ virtual void onHalLatencyModesChanged_l() {}
// see note at declaration of mStandby, mOutDevice and mInDevice
@@ -921,6 +932,8 @@
}
virtual void checkOutputStageEffects() {}
+ virtual void setHalLatencyMode_l() {}
+
void dumpInternals_l(int fd, const Vector<String16>& args) override;
void dumpTracks_l(int fd, const Vector<String16>& args) override;
@@ -1064,6 +1077,15 @@
bool hasMixer() const {
return mType == MIXER || mType == DUPLICATING || mType == SPATIALIZER;
}
+
+ virtual status_t setRequestedLatencyMode(
+ audio_latency_mode_t mode __unused) { return INVALID_OPERATION; }
+
+ virtual status_t getSupportedLatencyModes(
+ std::vector<audio_latency_mode_t>* modes __unused) {
+ return INVALID_OPERATION;
+ }
+
protected:
// updated by readOutputParameters_l()
size_t mNormalFrameCount; // normal mixer and effects
@@ -1707,7 +1729,8 @@
}
};
-class SpatializerThread : public MixerThread {
+class SpatializerThread : public MixerThread,
+ public StreamOutHalInterfaceLatencyModeCallback {
public:
SpatializerThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
@@ -1718,10 +1741,35 @@
bool hasFastMixer() const override { return false; }
+ status_t createAudioPatch_l(const struct audio_patch *patch,
+ audio_patch_handle_t *handle) override;
+
+ // RefBase
+ virtual void onFirstRef();
+
+ // StreamOutHalInterfaceLatencyModeCallback
+ void onRecommendedLatencyModeChanged(std::vector<audio_latency_mode_t> modes) override;
+
+ status_t setRequestedLatencyMode(audio_latency_mode_t mode) override;
+ status_t getSupportedLatencyModes(std::vector<audio_latency_mode_t>* modes) override;
+
protected:
void checkOutputStageEffects() override;
+ void onHalLatencyModesChanged_l() override;
+ void setHalLatencyMode_l() override;
private:
+ void updateHalSupportedLatencyModes_l();
+
+ // Support low latency mode by default as unless explicitly indicated by the audio HAL
+ // we assume the audio path is compatible with the head tracking latency requirements
+ std::vector<audio_latency_mode_t> mSupportedLatencyModes = {AUDIO_LATENCY_MODE_LOW};
+ // default to invalid value to force first update to the audio HAL
+ audio_latency_mode_t mSetLatencyMode =
+ (audio_latency_mode_t)AUDIO_LATENCY_MODE_INVALID;
+ // Do not request a specific mode by default
+ audio_latency_mode_t mRequestedLatencyMode = AUDIO_LATENCY_MODE_FREE;
+
sp<EffectHandle> mFinalDownMixer;
};
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 2677ab3..20bfbb0 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -397,6 +397,8 @@
int64_t mLogStartFrames = 0; // Timestamp frames at start()
double mLogLatencyMs = 0.; // Track the last log latency
+ bool mLogForceVolumeUpdate = true; // force volume update to TrackMetrics.
+
TrackMetrics mTrackMetrics;
bool mServerLatencySupported = false;
diff --git a/services/audioflinger/TrackMetrics.h b/services/audioflinger/TrackMetrics.h
index 30d69ab..6fc70d6 100644
--- a/services/audioflinger/TrackMetrics.h
+++ b/services/audioflinger/TrackMetrics.h
@@ -64,7 +64,6 @@
AMEDIAMETRICS_PROP_EVENT_VALUE_BEGINAUDIOINTERVALGROUP, devices.c_str());
}
++mIntervalCount;
- mIntervalStartTimeNs = systemTime();
}
void logConstructor(pid_t creatorPid, uid_t creatorUid, int32_t internalTrackId,
@@ -90,11 +89,9 @@
// Called when we are removed from the Thread.
void logEndInterval() {
std::lock_guard l(mLock);
- if (mIntervalStartTimeNs != 0) {
- const int64_t elapsedTimeNs = systemTime() - mIntervalStartTimeNs;
- mIntervalStartTimeNs = 0;
- mCumulativeTimeNs += elapsedTimeNs;
- mDeviceTimeNs += elapsedTimeNs;
+ if (mLastVolumeChangeTimeNs != 0) {
+ logVolume_l(mVolume); // flush out the last volume.
+ mLastVolumeChangeTimeNs = 0;
}
}
@@ -133,20 +130,8 @@
// may be called multiple times during an interval
void logVolume(float volume) {
- const int64_t timeNs = systemTime();
std::lock_guard l(mLock);
- if (mStartVolumeTimeNs == 0) {
- mDeviceVolume = mVolume = volume;
- mLastVolumeChangeTimeNs = mStartVolumeTimeNs = timeNs;
- updateMinMaxVolume(0, mVolume);
- return;
- }
- const int64_t durationNs = timeNs - mLastVolumeChangeTimeNs;
- updateMinMaxVolume(durationNs, mVolume);
- mDeviceVolume = (mDeviceVolume * (mLastVolumeChangeTimeNs - mStartVolumeTimeNs) +
- mVolume * durationNs) / (timeNs - mStartVolumeTimeNs);
- mVolume = volume;
- mLastVolumeChangeTimeNs = timeNs;
+ logVolume_l(volume);
}
// Use absolute numbers returned by AudioTrackShared.
@@ -158,6 +143,7 @@
}
private:
+
// no lock required - all arguments and constants.
void deliverDeviceMetrics(const char *eventName, const char *devices) const {
mediametrics::LogItem(mMetricsId)
@@ -167,6 +153,23 @@
.record();
}
+ void logVolume_l(float volume) REQUIRES(mLock) {
+ const int64_t timeNs = systemTime();
+ const int64_t durationNs = mLastVolumeChangeTimeNs == 0
+ ? 0 : timeNs - mLastVolumeChangeTimeNs;
+ if (durationNs > 0) {
+ // See West's algorithm for weighted averages
+ // https://en.wikipedia.org/wiki/Algorithms_for_calculating_variance
+ mDeviceVolume += (mVolume - mDeviceVolume) * durationNs
+ / (durationNs + mDeviceTimeNs);
+ mDeviceTimeNs += durationNs;
+ mCumulativeTimeNs += durationNs;
+ }
+ updateMinMaxVolume(durationNs, mVolume); // always update.
+ mVolume = volume;
+ mLastVolumeChangeTimeNs = timeNs;
+ }
+
void deliverCumulativeMetrics(const char *eventName) const REQUIRES(mLock) {
if (mIntervalCount > 0) {
mediametrics::LogItem item(mMetricsId);
@@ -199,14 +202,12 @@
// mDevices is not reset by resetIntervalGroupMetrics.
mIntervalCount = 0;
- mIntervalStartTimeNs = 0;
// mCumulativeTimeNs is not reset by resetIntervalGroupMetrics.
mDeviceTimeNs = 0;
mVolume = 0.f;
mDeviceVolume = 0.f;
- mStartVolumeTimeNs = 0;
- mLastVolumeChangeTimeNs = 0;
+ mLastVolumeChangeTimeNs = 0; // last time volume logged, cleared on endInterval
mMinVolume = AMEDIAMETRICS_INITIAL_MIN_VOLUME;
mMaxVolume = AMEDIAMETRICS_INITIAL_MAX_VOLUME;
mMinVolumeDurationNs = 0;
@@ -230,14 +231,12 @@
// Number of intervals and playing time
int32_t mIntervalCount GUARDED_BY(mLock) = 0;
- int64_t mIntervalStartTimeNs GUARDED_BY(mLock) = 0;
- int64_t mCumulativeTimeNs GUARDED_BY(mLock) = 0;
- int64_t mDeviceTimeNs GUARDED_BY(mLock) = 0;
+ int64_t mCumulativeTimeNs GUARDED_BY(mLock) = 0; // total time.
+ int64_t mDeviceTimeNs GUARDED_BY(mLock) = 0; // time on device.
// Average volume
- double mVolume GUARDED_BY(mLock) = 0.f;
- double mDeviceVolume GUARDED_BY(mLock) = 0.f;
- int64_t mStartVolumeTimeNs GUARDED_BY(mLock) = 0;
+ double mVolume GUARDED_BY(mLock) = 0.f; // last set volume.
+ double mDeviceVolume GUARDED_BY(mLock) = 0.f; // running average volume.
int64_t mLastVolumeChangeTimeNs GUARDED_BY(mLock) = 0;
// Min/Max volume
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 6135020..b422bb2 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1123,6 +1123,7 @@
.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
mLogLatencyMs = 0.;
}
+ mLogForceVolumeUpdate = true; // at least one volume logged for metrics when starting.
if (status == NO_ERROR || status == ALREADY_EXISTS) {
// for streaming tracks, remove the buffer read stop limit.
@@ -1394,12 +1395,21 @@
if (mFinalVolume != volume) { // Compare to an epsilon if too many meaningless updates
mFinalVolume = volume;
setMetadataHasChanged();
- mTrackMetrics.logVolume(volume);
+ mLogForceVolumeUpdate = true;
+ }
+ if (mLogForceVolumeUpdate) {
+ mLogForceVolumeUpdate = false;
+ mTrackMetrics.logVolume(mFinalVolume);
}
}
void AudioFlinger::PlaybackThread::Track::copyMetadataTo(MetadataInserter& backInserter) const
{
+ // Do not forward metadata for PatchTrack with unspecified stream type
+ if (mStreamType == AUDIO_STREAM_PATCH) {
+ return;
+ }
+
playback_track_metadata_v7_t metadata;
metadata.base = {
.usage = mAttr.usage,
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
index c2a20c6..bb1699e 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
@@ -70,8 +70,9 @@
* @return OK if the request is valid
* otherwise if the request is not supported
*/
- status_t getOutputForAttr(const audio_attributes_t& attributes, uid_t uid,
- audio_output_flags_t flags,
+ status_t getOutputForAttr(const audio_attributes_t& attributes,
+ const audio_config_base_t& config,
+ uid_t uid, audio_output_flags_t flags,
sp<AudioPolicyMix> &primaryMix,
std::vector<sp<AudioPolicyMix>> *secondaryMixes);
@@ -126,6 +127,7 @@
enum class MixMatchStatus { MATCH, NO_MATCH, INVALID_MIX };
MixMatchStatus mixMatch(const AudioMix* mix, size_t mixIndex,
const audio_attributes_t& attributes,
+ const audio_config_base_t& config,
uid_t uid);
};
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index 546f56b..551eab6 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -151,7 +151,7 @@
}
status_t AudioPolicyMixCollection::getOutputForAttr(
- const audio_attributes_t& attributes, uid_t uid,
+ const audio_attributes_t& attributes, const audio_config_base_t& config, uid_t uid,
audio_output_flags_t flags,
sp<AudioPolicyMix> &primaryMix,
std::vector<sp<AudioPolicyMix>> *secondaryMixes)
@@ -177,7 +177,7 @@
continue; // Primary output already found
}
- switch (mixMatch(policyMix.get(), i, attributes, uid)) {
+ switch (mixMatch(policyMix.get(), i, attributes, config, uid)) {
case MixMatchStatus::INVALID_MIX:
// The mix has contradictory rules, ignore it
// TODO: reject invalid mix at registration
@@ -202,7 +202,8 @@
}
AudioPolicyMixCollection::MixMatchStatus AudioPolicyMixCollection::mixMatch(
- const AudioMix* mix, size_t mixIndex, const audio_attributes_t& attributes, uid_t uid) {
+ const AudioMix* mix, size_t mixIndex, const audio_attributes_t& attributes,
+ const audio_config_base_t& config, uid_t uid) {
if (mix->mMixType == MIX_TYPE_PLAYERS) {
// Loopback render mixes are created from a public API and thus restricted
@@ -229,6 +230,14 @@
}
}
+ // Permit match only if requested format and mix format are PCM and can be format
+ // adapted by the mixer, or are the same (compressed) format.
+ if (!((audio_is_linear_pcm(config.format) && audio_is_linear_pcm(mix->mFormat.format)) ||
+ (config.format == mix->mFormat.format)) &&
+ config.format != AUDIO_CONFIG_BASE_INITIALIZER.format) {
+ return MixMatchStatus::NO_MATCH;
+ }
+
int userId = (int) multiuser_get_user_id(uid);
// TODO if adding more player rules (currently only 2), make rule handling "generic"
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
index d4d514d..08d6453 100644
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -195,6 +195,12 @@
AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,
AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, }));
}
+ // If connected to a dock, never use the device speaker for calls
+ if (!availableOutputDevices.getDevicesFromTypes({AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET})
+ .isEmpty()) {
+ availableOutputDevices.remove(
+ availableOutputDevices.getDevicesFromTypes({AUDIO_DEVICE_OUT_SPEAKER}));
+ }
} break;
case STRATEGY_ACCESSIBILITY: {
// do not route accessibility prompts to a digital output currently configured with a
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 744609f..912d53a 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1054,12 +1054,12 @@
if (curProfile->getFlags() != AUDIO_OUTPUT_FLAG_SPATIALIZER) {
continue;
}
- // reject profiles not corresponding to a device currently available
- DeviceVector supportedDevices = curProfile->getSupportedDevices();
- if (!mAvailableOutputDevices.containsAtLeastOne(supportedDevices)) {
- continue;
- }
if (!devices.empty()) {
+ // reject profiles not corresponding to a device currently available
+ DeviceVector supportedDevices = curProfile->getSupportedDevices();
+ if (!mAvailableOutputDevices.containsAtLeastOne(supportedDevices)) {
+ continue;
+ }
if (supportedDevices.getDevicesFromDeviceTypeAddrVec(devices).size()
!= devices.size()) {
continue;
@@ -1161,7 +1161,12 @@
// otherwise, fallback to the dynamic policies, if none match, query the engine.
// Secondary outputs are always found by dynamic policies as the engine do not support them
sp<AudioPolicyMix> primaryMix;
- status = mPolicyMixes.getOutputForAttr(*resultAttr, uid, *flags, primaryMix, secondaryMixes);
+ const audio_config_base_t clientConfig = {.sample_rate = config->sample_rate,
+ .channel_mask = config->channel_mask,
+ .format = config->format,
+ };
+ status = mPolicyMixes.getOutputForAttr(*resultAttr, clientConfig, uid, *flags, primaryMix,
+ secondaryMixes);
if (status != OK) {
return status;
}
@@ -1170,10 +1175,9 @@
bool usePrimaryOutputFromPolicyMixes = requestedDevice == nullptr && primaryMix != nullptr;
// FIXME: in case of RENDER policy, the output capabilities should be checked
- if ((usePrimaryOutputFromPolicyMixes
- || (secondaryMixes != nullptr && !secondaryMixes->empty()))
- && !audio_is_linear_pcm(config->format)) {
- ALOGD("%s: rejecting request as dynamic audio policy only support pcm", __func__);
+ if ((secondaryMixes != nullptr && !secondaryMixes->empty())
+ && !audio_is_linear_pcm(config->format)) {
+ ALOGD("%s: rejecting request as secondary mixes only support pcm", __func__);
return BAD_VALUE;
}
if (usePrimaryOutputFromPolicyMixes) {
@@ -1182,19 +1186,27 @@
primaryMix->mDeviceAddress,
AUDIO_FORMAT_DEFAULT);
sp<SwAudioOutputDescriptor> policyDesc = primaryMix->getOutput();
- if (deviceDesc != nullptr
- && (policyDesc == nullptr || (policyDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT))) {
+ bool tryDirectForFlags = policyDesc == nullptr ||
+ (policyDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT);
+ // if a direct output can be opened to deliver the track's multi-channel content to the
+ // output rather than being downmixed by the primary output, then use this direct
+ // output by by-passing the primary mix if possible, otherwise fall-through to primary
+ // mix.
+ bool tryDirectForChannelMask = policyDesc != nullptr
+ && (audio_channel_count_from_out_mask(policyDesc->getConfig().channel_mask) <
+ audio_channel_count_from_out_mask(config->channel_mask));
+ if (deviceDesc != nullptr && (tryDirectForFlags || tryDirectForChannelMask)) {
audio_io_handle_t newOutput;
status = openDirectOutput(
*stream, session, config,
(audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT),
DeviceVector(deviceDesc), &newOutput);
- if (status != NO_ERROR) {
- policyDesc = nullptr;
- } else {
+ if (status == NO_ERROR) {
policyDesc = mOutputs.valueFor(newOutput);
primaryMix->setOutput(policyDesc);
- }
+ } else if (tryDirectForFlags) {
+ policyDesc = nullptr;
+ } // otherwise use primary if available.
}
if (policyDesc != nullptr) {
policyDesc->mPolicyMix = primaryMix;
@@ -1868,7 +1880,8 @@
// (see b/200293124, the incorrect selection of AUDIO_OUTPUT_FLAG_VOIP_RX).
// 3: the output supporting the exact channel mask
// 4: the output with a higher channel count than requested
- // 5: the output with a higher sampling rate than requested
+ // 5: the output with the highest sampling rate if the requested sample rate is
+ // greater than default sampling rate
// 6: the output with the highest number of requested performance flags
// 7: the output with the bit depth the closest to the requested one
// 8: the primary output
@@ -1928,8 +1941,7 @@
}
// sampling rate match
- if (samplingRate > SAMPLE_RATE_HZ_DEFAULT &&
- samplingRate <= outputDesc->getSamplingRate()) {
+ if (samplingRate > SAMPLE_RATE_HZ_DEFAULT) {
currentMatchCriteria[4] = outputDesc->getSamplingRate();
}
@@ -5353,10 +5365,6 @@
}
}
- // The caller can have the devices criteria ignored by passing and empty vector, and
- // getSpatializerOutputProfile() will ignore the devices when looking for a match.
- // Otherwise an output profile supporting a spatializer effect that can be routed
- // to the specified devices must exist.
sp<IOProfile> profile =
getSpatializerOutputProfile(config, devices);
if (profile == nullptr) {
@@ -6330,8 +6338,8 @@
continue;
}
sp<AudioPolicyMix> primaryMix;
- status_t status = mPolicyMixes.getOutputForAttr(client->attributes(), client->uid(),
- client->flags(), primaryMix, nullptr);
+ status_t status = mPolicyMixes.getOutputForAttr(client->attributes(), client->config(),
+ client->uid(), client->flags(), primaryMix, nullptr);
if (status != OK) {
continue;
}
@@ -6443,8 +6451,8 @@
for (const sp<TrackClientDescriptor>& client : outputDescriptor->getClientIterable()) {
sp<AudioPolicyMix> primaryMix;
std::vector<sp<AudioPolicyMix>> secondaryMixes;
- status_t status = mPolicyMixes.getOutputForAttr(client->attributes(), client->uid(),
- client->flags(), primaryMix, &secondaryMixes);
+ status_t status = mPolicyMixes.getOutputForAttr(client->attributes(), client->config(),
+ client->uid(), client->flags(), primaryMix, &secondaryMixes);
std::vector<sp<SwAudioOutputDescriptor>> secondaryDescs;
for (auto &secondaryMix : secondaryMixes) {
sp<SwAudioOutputDescriptor> outputDesc = secondaryMix->getOutput();
@@ -6702,8 +6710,8 @@
// check dynamic policies but only for primary descriptors (secondary not used for audible
// audio routing, only used for duplication for playback capture)
sp<AudioPolicyMix> policyMix;
- status_t status = mPolicyMixes.getOutputForAttr(attr, 0 /*uid unknown here*/,
- AUDIO_OUTPUT_FLAG_NONE, policyMix, nullptr /* secondaryMixes */);
+ status_t status = mPolicyMixes.getOutputForAttr(attr, AUDIO_CONFIG_BASE_INITIALIZER,
+ 0 /*uid unknown here*/, AUDIO_OUTPUT_FLAG_NONE, policyMix, nullptr);
if (status != OK) {
return status;
}
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index db0ee15..87e6974 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -1099,6 +1099,18 @@
const audio_config_t *config,
const AudioDeviceTypeAddrVector &devices) const;
+
+ /**
+ * @brief Gets an IOProfile for a spatializer output with the best match with
+ * provided arguments.
+ * The caller can have the devices criteria ignored by passing and empty vector, and
+ * getSpatializerOutputProfile() will ignore the devices when looking for a match.
+ * Otherwise an output profile supporting a spatializer effect that can be routed
+ * to the specified devices must exist.
+ * @param config audio configuration describing the audio format, channels, sample rate...
+ * @param devices the sink audio device selected for playback
+ * @return an IOProfile that canbe used to open a spatializer output.
+ */
sp<IOProfile> getSpatializerOutputProfile(const audio_config_t *config,
const AudioDeviceTypeAddrVector &devices) const;
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index e7d945f..691f13c 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -516,8 +516,6 @@
void AudioPolicyService::doOnCheckSpatializer()
{
- Mutex::Autolock _l(mLock);
-
ALOGI("%s mSpatializer %p level %d", __func__, mSpatializer.get(), (int)mSpatializer->getLevel());
if (mSpatializer != nullptr) {
@@ -527,6 +525,8 @@
audio_io_handle_t newOutput;
const audio_attributes_t attr = attributes_initializer(AUDIO_USAGE_MEDIA);
audio_config_base_t config = mSpatializer->getAudioInConfig();
+
+ Mutex::Autolock _l(mLock);
status_t status =
mAudioPolicyManager->getSpatializerOutput(&config, &attr, &newOutput);
ALOGV("%s currentOutput %d newOutput %d channel_mask %#x",
@@ -538,21 +538,19 @@
mLock.unlock();
// It is OK to call detachOutput() is none is already attached.
mSpatializer->detachOutput();
- if (status != NO_ERROR || newOutput == AUDIO_IO_HANDLE_NONE) {
- mLock.lock();
- return;
+ if (status == NO_ERROR && newOutput != AUDIO_IO_HANDLE_NONE) {
+ status = mSpatializer->attachOutput(newOutput, numActiveTracks);
}
- status = mSpatializer->attachOutput(newOutput, numActiveTracks);
mLock.lock();
if (status != NO_ERROR) {
mAudioPolicyManager->releaseSpatializerOutput(newOutput);
}
} else if (mSpatializer->getLevel() == media::SpatializationLevel::NONE
&& mSpatializer->getOutput() != AUDIO_IO_HANDLE_NONE) {
- mLock.unlock();
audio_io_handle_t output = mSpatializer->detachOutput();
- mLock.lock();
+
if (output != AUDIO_IO_HANDLE_NONE) {
+ Mutex::Autolock _l(mLock);
mAudioPolicyManager->releaseSpatializerOutput(output);
}
}
@@ -581,19 +579,16 @@
void AudioPolicyService::doOnUpdateActiveSpatializerTracks()
{
- sp<Spatializer> spatializer;
+ if (mSpatializer == nullptr) {
+ return;
+ }
+ audio_io_handle_t output = mSpatializer->getOutput();
size_t activeClients;
{
Mutex::Autolock _l(mLock);
- if (mSpatializer == nullptr) {
- return;
- }
- spatializer = mSpatializer;
- activeClients = countActiveClientsOnOutput_l(mSpatializer->getOutput());
+ activeClients = countActiveClientsOnOutput_l(output);
}
- if (spatializer != nullptr) {
- spatializer->updateActiveTracks(activeClients);
- }
+ mSpatializer->updateActiveTracks(activeClients);
}
status_t AudioPolicyService::clientCreateAudioPatch(const struct audio_patch *patch,
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 3a08cf8..a87d871 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -1060,6 +1060,7 @@
CaptureStateNotifier mCaptureStateNotifier;
+ // created in onFirstRef() and never cleared: does not need to be guarded by mLock
sp<Spatializer> mSpatializer;
void *mLibraryHandle = nullptr;
diff --git a/services/audiopolicy/service/Spatializer.cpp b/services/audiopolicy/service/Spatializer.cpp
index c199a76..9f0a901 100644
--- a/services/audiopolicy/service/Spatializer.cpp
+++ b/services/audiopolicy/service/Spatializer.cpp
@@ -84,6 +84,7 @@
kWhatOnFramesProcessed, // AudioEffect::EVENT_FRAMES_PROCESSED
kWhatOnHeadToStagePose, // SpatializerPoseController::Listener::onHeadToStagePose
kWhatOnActualModeChange, // SpatializerPoseController::Listener::onActualModeChange
+ kWhatOnLatencyModesChanged, // Spatializer::onSupportedLatencyModesChanged
};
static constexpr const char *kNumFramesKey = "numFrames";
static constexpr const char *kModeKey = "mode";
@@ -93,15 +94,27 @@
static constexpr const char *kRotation0Key = "rotation0";
static constexpr const char *kRotation1Key = "rotation1";
static constexpr const char *kRotation2Key = "rotation2";
+ static constexpr const char *kLatencyModesKey = "latencyModes";
+
+ class LatencyModes : public RefBase {
+ public:
+ LatencyModes(audio_io_handle_t output,
+ const std::vector<audio_latency_mode_t>& latencyModes)
+ : mOutput(output), mLatencyModes(latencyModes) {}
+ ~LatencyModes() = default;
+
+ audio_io_handle_t mOutput;
+ std::vector<audio_latency_mode_t> mLatencyModes;
+ };
void onMessageReceived(const sp<AMessage> &msg) override {
+ sp<Spatializer> spatializer = mSpatializer.promote();
+ if (spatializer == nullptr) {
+ ALOGW("%s: Cannot promote spatializer", __func__);
+ return;
+ }
switch (msg->what()) {
case kWhatOnFramesProcessed: {
- sp<Spatializer> spatializer = mSpatializer.promote();
- if (spatializer == nullptr) {
- ALOGW("%s: Cannot promote spatializer", __func__);
- return;
- }
int numFrames;
if (!msg->findInt32(kNumFramesKey, &numFrames)) {
ALOGE("%s: Cannot find num frames!", __func__);
@@ -112,11 +125,6 @@
}
} break;
case kWhatOnHeadToStagePose: {
- sp<Spatializer> spatializer = mSpatializer.promote();
- if (spatializer == nullptr) {
- ALOGW("%s: Cannot promote spatializer", __func__);
- return;
- }
std::vector<float> headToStage(sHeadPoseKeys.size());
for (size_t i = 0 ; i < sHeadPoseKeys.size(); i++) {
if (!msg->findFloat(sHeadPoseKeys[i], &headToStage[i])) {
@@ -127,18 +135,25 @@
spatializer->onHeadToStagePoseMsg(headToStage);
} break;
case kWhatOnActualModeChange: {
- sp<Spatializer> spatializer = mSpatializer.promote();
- if (spatializer == nullptr) {
- ALOGW("%s: Cannot promote spatializer", __func__);
- return;
- }
int mode;
- if (!msg->findInt32(EngineCallbackHandler::kModeKey, &mode)) {
+ if (!msg->findInt32(kModeKey, &mode)) {
ALOGE("%s: Cannot find actualMode!", __func__);
return;
}
spatializer->onActualModeChangeMsg(static_cast<HeadTrackingMode>(mode));
} break;
+
+ case kWhatOnLatencyModesChanged: {
+ sp<RefBase> object;
+ if (!msg->findObject(kLatencyModesKey, &object)) {
+ ALOGE("%s: Cannot find latency modes!", __func__);
+ return;
+ }
+ sp<LatencyModes> latencyModes = static_cast<LatencyModes*>(object.get());
+ spatializer->onSupportedLatencyModesChangedMsg(
+ latencyModes->mOutput, std::move(latencyModes->mLatencyModes));
+ } break;
+
default:
LOG_ALWAYS_FATAL("Invalid callback message %d", msg->what());
}
@@ -714,7 +729,9 @@
mEngine->setEnabled(false);
mEngine.clear();
mPoseController.reset();
+ AudioSystem::removeSupportedLatencyModesCallback(this);
}
+
// create FX instance on output
AttributionSourceState attributionSource = AttributionSourceState();
mEngine = new AudioEffect(attributionSource);
@@ -730,6 +747,13 @@
outputChanged = mOutput != output;
mOutput = output;
mNumActiveTracks = numActiveTracks;
+ AudioSystem::addSupportedLatencyModesCallback(this);
+
+ std::vector<audio_latency_mode_t> latencyModes;
+ status = AudioSystem::getSupportedLatencyModes(mOutput, &latencyModes);
+ if (status == OK) {
+ mSupportedLatencyModes = latencyModes;
+ }
checkEngineState_l();
if (mSupportsHeadTracking) {
@@ -759,6 +783,7 @@
// remove FX instance
mEngine->setEnabled(false);
mEngine.clear();
+ AudioSystem::removeSupportedLatencyModesCallback(this);
output = mOutput;
mOutput = AUDIO_IO_HANDLE_NONE;
mPoseController.reset();
@@ -771,6 +796,27 @@
return output;
}
+void Spatializer::onSupportedLatencyModesChanged(
+ audio_io_handle_t output, const std::vector<audio_latency_mode_t>& modes) {
+ ALOGV("%s output %d num modes %zu", __func__, (int)output, modes.size());
+ sp<AMessage> msg =
+ new AMessage(EngineCallbackHandler::kWhatOnLatencyModesChanged, mHandler);
+ msg->setObject(EngineCallbackHandler::kLatencyModesKey,
+ sp<EngineCallbackHandler::LatencyModes>::make(output, modes));
+ msg->post();
+}
+
+void Spatializer::onSupportedLatencyModesChangedMsg(
+ audio_io_handle_t output, std::vector<audio_latency_mode_t>&& modes) {
+ std::lock_guard lock(mLock);
+ ALOGV("%s output %d mOutput %d num modes %zu",
+ __func__, (int)output, (int)mOutput, modes.size());
+ if (output == mOutput) {
+ mSupportedLatencyModes = std::move(modes);
+ checkSensorsState_l();
+ }
+}
+
void Spatializer::updateActiveTracks(size_t numActiveTracks) {
std::lock_guard lock(mLock);
if (mNumActiveTracks != numActiveTracks) {
@@ -781,17 +827,25 @@
}
void Spatializer::checkSensorsState_l() {
+ audio_latency_mode_t requestedLatencyMode = AUDIO_LATENCY_MODE_FREE;
+ bool lowLatencySupported = mSupportedLatencyModes.empty()
+ || (std::find(mSupportedLatencyModes.begin(), mSupportedLatencyModes.end(),
+ AUDIO_LATENCY_MODE_LOW) != mSupportedLatencyModes.end());
if (mSupportsHeadTracking && mPoseController != nullptr) {
- if (mNumActiveTracks > 0 && mLevel != SpatializationLevel::NONE
+ if (lowLatencySupported && mNumActiveTracks > 0 && mLevel != SpatializationLevel::NONE
&& mDesiredHeadTrackingMode != HeadTrackingMode::STATIC
&& mHeadSensor != SpatializerPoseController::INVALID_SENSOR) {
mPoseController->setHeadSensor(mHeadSensor);
mPoseController->setScreenSensor(mScreenSensor);
+ requestedLatencyMode = AUDIO_LATENCY_MODE_LOW;
} else {
mPoseController->setHeadSensor(SpatializerPoseController::INVALID_SENSOR);
mPoseController->setScreenSensor(SpatializerPoseController::INVALID_SENSOR);
}
}
+ if (mOutput != AUDIO_IO_HANDLE_NONE) {
+ AudioSystem::setRequestedLatencyMode(mOutput, requestedLatencyMode);
+ }
}
void Spatializer::checkEngineState_l() {
diff --git a/services/audiopolicy/service/Spatializer.h b/services/audiopolicy/service/Spatializer.h
index 29f4b08..a7d9f32 100644
--- a/services/audiopolicy/service/Spatializer.h
+++ b/services/audiopolicy/service/Spatializer.h
@@ -85,7 +85,8 @@
*/
class Spatializer : public media::BnSpatializer,
public IBinder::DeathRecipient,
- private SpatializerPoseController::Listener {
+ private SpatializerPoseController::Listener,
+ public virtual AudioSystem::SupportedLatencyModesCallback {
public:
static sp<Spatializer> create(SpatializerPolicyCallback *callback);
@@ -122,6 +123,10 @@
/** IBinder::DeathRecipient. Listen to the death of the INativeSpatializerCallback. */
virtual void binderDied(const wp<IBinder>& who);
+ /** SupportedLatencyModesCallback */
+ void onSupportedLatencyModesChanged(
+ audio_io_handle_t output, const std::vector<audio_latency_mode_t>& modes) override;
+
/** Registers a INativeSpatializerCallback when a client is attached to this Spatializer
* by audio policy service.
*/
@@ -162,6 +167,8 @@
void onHeadToStagePoseMsg(const std::vector<float>& headToStage);
void onActualModeChangeMsg(media::HeadTrackingMode mode);
+ void onSupportedLatencyModesChangedMsg(
+ audio_io_handle_t output, std::vector<audio_latency_mode_t>&& modes);
static constexpr int kMaxEffectParamValues = 10;
/**
@@ -354,6 +361,7 @@
sp<EngineCallbackHandler> mHandler;
size_t mNumActiveTracks GUARDED_BY(mLock) = 0;
+ std::vector<audio_latency_mode_t> mSupportedLatencyModes GUARDED_BY(mLock);
static const std::vector<const char *> sHeadPoseKeys;
};
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 5429176..43b1a2a 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -148,6 +148,8 @@
std::unique_ptr<AudioPolicyManagerTestClient> mClient;
std::unique_ptr<AudioPolicyTestManager> mManager;
+
+ const uint32_t k48000SamplingRate = 48000;
};
void AudioPolicyManagerTest::SetUp() {
@@ -414,11 +416,11 @@
AudioPolicyConfig& config = mManager->getConfig();
mMsdOutputDevice = new DeviceDescriptor(AUDIO_DEVICE_OUT_BUS);
sp<AudioProfile> pcmOutputProfile = new AudioProfile(
- AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000);
+ AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, k48000SamplingRate);
sp<AudioProfile> ac3OutputProfile = new AudioProfile(
- AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000);
+ AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, k48000SamplingRate);
sp<AudioProfile> iec958OutputProfile = new AudioProfile(
- AUDIO_FORMAT_IEC60958, AUDIO_CHANNEL_OUT_STEREO, 48000);
+ AUDIO_FORMAT_IEC60958, AUDIO_CHANNEL_OUT_STEREO, k48000SamplingRate);
mMsdOutputDevice->addAudioProfile(pcmOutputProfile);
mMsdOutputDevice->addAudioProfile(ac3OutputProfile);
mMsdOutputDevice->addAudioProfile(iec958OutputProfile);
@@ -473,7 +475,7 @@
// Add a profile with another encoding to the default device to test routing
// of streams that are not supported by MSD.
sp<AudioProfile> dtsOutputProfile = new AudioProfile(
- AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, 48000);
+ AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, k48000SamplingRate);
config.getDefaultOutputDevice()->addAudioProfile(dtsOutputProfile);
sp<OutputProfile> primaryEncodedOutputProfile = new OutputProfile("encoded");
primaryEncodedOutputProfile->addAudioProfile(dtsOutputProfile);
@@ -491,7 +493,7 @@
// Add HDMI input device with IEC60958 profile for HDMI in -> MSD patching.
mHdmiInputDevice = new DeviceDescriptor(AUDIO_DEVICE_IN_HDMI);
sp<AudioProfile> iec958InputProfile = new AudioProfile(
- AUDIO_FORMAT_IEC60958, AUDIO_CHANNEL_IN_STEREO, 48000);
+ AUDIO_FORMAT_IEC60958, AUDIO_CHANNEL_IN_STEREO, k48000SamplingRate);
mHdmiInputDevice->addAudioProfile(iec958InputProfile);
config.addDevice(mHdmiInputDevice);
sp<InputProfile> hdmiInputProfile = new InputProfile("hdmi input");
@@ -556,8 +558,8 @@
TEST_P(AudioPolicyManagerTestMsd, GetOutputForAttrEncodedRoutesToMsd) {
const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
- getOutputForAttr(&selectedDeviceId,
- AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1,
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(mExpectedAudioPatchCount, patchCount.deltaFromSnapshot());
}
@@ -566,7 +568,7 @@
const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId,
- AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000);
+ AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, k48000SamplingRate);
ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(mExpectedAudioPatchCount, patchCount.deltaFromSnapshot());
}
@@ -574,13 +576,13 @@
TEST_P(AudioPolicyManagerTestMsd, GetOutputForAttrEncodedPlusPcmRoutesToMsd) {
const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
- getOutputForAttr(&selectedDeviceId,
- AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1,
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(mExpectedAudioPatchCount, patchCount.deltaFromSnapshot());
selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId,
- AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000);
+ AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, k48000SamplingRate);
ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(mExpectedAudioPatchCount, patchCount.deltaFromSnapshot());
}
@@ -588,8 +590,8 @@
TEST_P(AudioPolicyManagerTestMsd, GetOutputForAttrUnsupportedFormatBypassesMsd) {
const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
- getOutputForAttr(&selectedDeviceId,
- AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1,
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_NE(selectedDeviceId, mMsdOutputDevice->getId());
ASSERT_EQ(0, patchCount.deltaFromSnapshot());
}
@@ -600,9 +602,8 @@
const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
audio_port_handle_t portId;
- getOutputForAttr(&selectedDeviceId,
- AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT,
- nullptr /*output*/, &portId);
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1,
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_DIRECT, nullptr /*output*/, &portId);
ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(mExpectedAudioPatchCount, patchCount.deltaFromSnapshot());
mManager->releaseOutput(portId);
@@ -612,9 +613,8 @@
const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
audio_port_handle_t portId;
- getOutputForAttr(&selectedDeviceId,
- AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT,
- nullptr /*output*/, &portId);
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1,
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_DIRECT, nullptr /*output*/, &portId);
ASSERT_NE(selectedDeviceId, mMsdOutputDevice->getId());
ASSERT_EQ(-static_cast<int>(mExpectedAudioPatchCount), patchCount.deltaFromSnapshot());
mManager->releaseOutput(portId);
@@ -623,8 +623,8 @@
{
const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
- getOutputForAttr(&selectedDeviceId,
- AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1,
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(0, patchCount.deltaFromSnapshot());
}
@@ -653,8 +653,8 @@
ASSERT_EQ(AUDIO_FORMAT_IEC60958, patch->mPatch.sinks[0].format);
ASSERT_EQ(AUDIO_CHANNEL_IN_STEREO, patch->mPatch.sources[0].channel_mask);
ASSERT_EQ(AUDIO_CHANNEL_OUT_STEREO, patch->mPatch.sinks[0].channel_mask);
- ASSERT_EQ(48000, patch->mPatch.sources[0].sample_rate);
- ASSERT_EQ(48000, patch->mPatch.sinks[0].sample_rate);
+ ASSERT_EQ(k48000SamplingRate, patch->mPatch.sources[0].sample_rate);
+ ASSERT_EQ(k48000SamplingRate, patch->mPatch.sinks[0].sample_rate);
ASSERT_EQ(1, patchCount.deltaFromSnapshot());
}
@@ -996,7 +996,7 @@
clearPolicyMix();
audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
- audioConfig.sample_rate = 48000;
+ audioConfig.sample_rate = k48000SamplingRate;
ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
AUDIO_DEVICE_OUT_REMOTE_SUBMIX, mMixAddress, audioConfig,
std::vector<PolicyMixTuple>());
@@ -1035,7 +1035,7 @@
audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
- audioConfig.sample_rate = 48000;
+ audioConfig.sample_rate = k48000SamplingRate;
ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
AUDIO_DEVICE_OUT_REMOTE_SUBMIX, mMixAddress, audioConfig,
std::vector<PolicyMixTuple>());
@@ -1273,7 +1273,7 @@
audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
- audioConfig.sample_rate = 48000;
+ audioConfig.sample_rate = k48000SamplingRate;
ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
AUDIO_DEVICE_OUT_REMOTE_SUBMIX, "", audioConfig, std::vector<PolicyMixTuple>());
ASSERT_EQ(INVALID_OPERATION, ret);
@@ -1307,7 +1307,7 @@
audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
- audioConfig.sample_rate = 48000;
+ audioConfig.sample_rate = k48000SamplingRate;
status_t ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
AUDIO_DEVICE_OUT_REMOTE_SUBMIX, mMixAddress, audioConfig, mUsageRules);
ASSERT_EQ(NO_ERROR, ret);
@@ -1323,7 +1323,7 @@
std::string tags = "addr=" + mMixAddress;
strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
getInputForAttr(attr, mTracker->getRiid(), &selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT,
- AUDIO_CHANNEL_IN_STEREO, 48000 /*sampleRate*/, AUDIO_INPUT_FLAG_NONE, &mPortId);
+ AUDIO_CHANNEL_IN_STEREO, k48000SamplingRate, AUDIO_INPUT_FLAG_NONE, &mPortId);
ASSERT_EQ(NO_ERROR, mManager->startInput(mPortId));
ASSERT_EQ(extractionPort.id, selectedDeviceId);
@@ -1350,8 +1350,8 @@
audio_port_handle_t playbackRoutedPortId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&playbackRoutedPortId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
- 48000 /*sampleRate*/, AUDIO_OUTPUT_FLAG_NONE,
- nullptr /*output*/, nullptr /*portId*/, attr);
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_NONE, nullptr /*output*/, nullptr /*portId*/,
+ attr);
if (std::find_if(begin(mUsageRules), end(mUsageRules), [&usage](const auto &usageRule) {
return (std::get<0>(usageRule) == usage) &&
(std::get<2>(usageRule) == RULE_MATCH_ATTRIBUTE_USAGE);}) != end(mUsageRules) ||
@@ -1493,7 +1493,7 @@
audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
audioConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
- audioConfig.sample_rate = 48000;
+ audioConfig.sample_rate = k48000SamplingRate;
status_t ret = addPolicyMix(MIX_TYPE_RECORDERS, MIX_ROUTE_FLAG_LOOP_BACK,
AUDIO_DEVICE_IN_REMOTE_SUBMIX, mMixAddress, audioConfig, mSourceRules);
ASSERT_EQ(NO_ERROR, ret);
@@ -1509,7 +1509,7 @@
std::string tags = std::string("addr=") + mMixAddress;
strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
- 48000 /*sampleRate*/, AUDIO_OUTPUT_FLAG_NONE, nullptr /*output*/, &mPortId, attr);
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_NONE, nullptr /*output*/, &mPortId, attr);
ASSERT_EQ(NO_ERROR, mManager->startOutput(mPortId));
ASSERT_EQ(injectionPort.id, getDeviceIdFromPatch(mClient->getLastAddedPatch()));
@@ -1537,7 +1537,7 @@
audio_port_handle_t captureRoutedPortId = AUDIO_PORT_HANDLE_NONE;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
getInputForAttr(attr, mTracker->getRiid(), &captureRoutedPortId, AUDIO_FORMAT_PCM_16_BIT,
- AUDIO_CHANNEL_IN_STEREO, 48000 /*sampleRate*/, AUDIO_INPUT_FLAG_NONE, &portId);
+ AUDIO_CHANNEL_IN_STEREO, k48000SamplingRate, AUDIO_INPUT_FLAG_NONE, &portId);
if (std::find_if(begin(mSourceRules), end(mSourceRules), [&source](const auto &sourceRule) {
return (std::get<1>(sourceRule) == source) &&
(std::get<2>(sourceRule) == RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET);})
@@ -1687,11 +1687,11 @@
// Try start input or output according to the device type
if (audio_is_output_devices(type)) {
getOutputForAttr(&routedPortId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
- 48000 /*sampleRate*/, AUDIO_OUTPUT_FLAG_NONE);
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_NONE);
} else if (audio_is_input_device(type)) {
RecordingActivityTracker tracker;
getInputForAttr({}, tracker.getRiid(), &routedPortId, AUDIO_FORMAT_PCM_16_BIT,
- AUDIO_CHANNEL_IN_STEREO, 48000 /*sampleRate*/, AUDIO_INPUT_FLAG_NONE);
+ AUDIO_CHANNEL_IN_STEREO, k48000SamplingRate, AUDIO_INPUT_FLAG_NONE);
}
ASSERT_EQ(devicePort.id, routedPortId);
@@ -1715,6 +1715,57 @@
)
);
+class AudioPolicyManagerCarTest : public AudioPolicyManagerTestDynamicPolicy {
+protected:
+ std::string getConfigFile() override { return sCarConfig; }
+
+ static const std::string sCarConfig;
+};
+
+const std::string AudioPolicyManagerCarTest::sCarConfig =
+ AudioPolicyManagerCarTest::sExecutableDir + "test_car_ap_atmos_offload_configuration.xml";
+
+TEST_F(AudioPolicyManagerCarTest, InitSuccess) {
+ // SetUp must finish with no assertions.
+}
+
+TEST_F(AudioPolicyManagerCarTest, Dump) {
+ dumpToLog();
+}
+
+TEST_F(AudioPolicyManagerCarTest, GetOutputForAttrAtmosOutputAfterRegisteringPolicyMix) {
+ status_t ret;
+ audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+ const std::string kTestBusMediaOutput = "bus0_media_out";
+ ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_RENDER,
+ AUDIO_DEVICE_OUT_BUS, kTestBusMediaOutput, audioConfig, std::vector<PolicyMixTuple>());
+ ASSERT_EQ(NO_ERROR, ret);
+
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_io_handle_t output;
+ audio_port_handle_t portId;
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_E_AC3_JOC, AUDIO_CHANNEL_OUT_5POINT1,
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_DIRECT, &output, &portId);
+ ASSERT_NE(AUDIO_PORT_HANDLE_NONE, selectedDeviceId);
+ sp<SwAudioOutputDescriptor> outDesc = mManager->getOutputs().valueFor(output);
+ ASSERT_NE(nullptr, outDesc.get());
+ ASSERT_EQ(AUDIO_FORMAT_E_AC3_JOC, outDesc->getFormat());
+ ASSERT_EQ(AUDIO_CHANNEL_OUT_5POINT1, outDesc->getChannelMask());
+ ASSERT_EQ(k48000SamplingRate, outDesc->getSamplingRate());
+
+ selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ output = AUDIO_IO_HANDLE_NONE;
+ portId = AUDIO_PORT_HANDLE_NONE;
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_7POINT1POINT4,
+ k48000SamplingRate, AUDIO_OUTPUT_FLAG_DIRECT, &output, &portId);
+ ASSERT_NE(AUDIO_PORT_HANDLE_NONE, selectedDeviceId);
+ outDesc = mManager->getOutputs().valueFor(output);
+ ASSERT_NE(nullptr, outDesc.get());
+ ASSERT_EQ(AUDIO_FORMAT_PCM_16_BIT, outDesc->getFormat());
+ ASSERT_EQ(AUDIO_CHANNEL_OUT_7POINT1POINT4, outDesc->getChannelMask());
+ ASSERT_EQ(k48000SamplingRate, outDesc->getSamplingRate());
+}
+
class AudioPolicyManagerTVTest : public AudioPolicyManagerTestWithConfigurationFile {
protected:
std::string getConfigFile() override { return sTvConfig; }
@@ -1735,8 +1786,8 @@
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
audio_io_handle_t output;
audio_port_handle_t portId;
- getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000,
- flags, &output, &portId);
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ k48000SamplingRate, flags, &output, &portId);
sp<SwAudioOutputDescriptor> outDesc = mManager->getOutputs().valueFor(output);
ASSERT_NE(nullptr, outDesc.get());
audio_port_v7 port = {};
diff --git a/services/audiopolicy/tests/resources/Android.bp b/services/audiopolicy/tests/resources/Android.bp
index ff4d568..5e71210 100644
--- a/services/audiopolicy/tests/resources/Android.bp
+++ b/services/audiopolicy/tests/resources/Android.bp
@@ -12,6 +12,7 @@
srcs: [
"test_audio_policy_configuration.xml",
"test_audio_policy_primary_only_configuration.xml",
+ "test_car_ap_atmos_offload_configuration.xml",
"test_invalid_audio_policy_configuration.xml",
"test_tv_apm_configuration.xml",
"test_settop_box_surround_configuration.xml",
diff --git a/services/audiopolicy/tests/resources/test_car_ap_atmos_offload_configuration.xml b/services/audiopolicy/tests/resources/test_car_ap_atmos_offload_configuration.xml
new file mode 100644
index 0000000..d131ed8
--- /dev/null
+++ b/services/audiopolicy/tests/resources/test_car_ap_atmos_offload_configuration.xml
@@ -0,0 +1,308 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2018 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <globalConfiguration speaker_drc_enabled="true"/>
+
+ <modules>
+ <!-- Primary Audio HAL -->
+ <module name="primary" halVersion="3.0">
+ <attachedDevices>
+ <!-- One bus per context -->
+ <item>bus0_media_out</item>
+ <item>bus1_navigation_out</item>
+ <item>bus2_voice_command_out</item>
+ <item>bus3_call_ring_out</item>
+ <item>bus4_call_out</item>
+ <item>bus5_alarm_out</item>
+ <item>bus6_notification_out</item>
+ <item>bus7_system_sound_out</item>
+ <!-- names with _audio_zone_# are used for defined an emulator rear seat audio zone
+ where each number # is the zone id number -->
+ <item>bus100_audio_zone_1</item>
+ <item>bus200_audio_zone_2</item>
+ <item>Built-In Mic</item>
+ <item>Built-In Back Mic</item>
+ <item>Echo-Reference Mic</item>
+ <item>FM Tuner</item>
+ <item>Tone Generator 0</item>
+ <item>Tone Generator 1</item>
+ </attachedDevices>
+ <defaultOutputDevice>bus0_media_out</defaultOutputDevice>
+ <mixPorts>
+ <mixPort name="mixport_bus0_media_out" role="source"
+ flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bus0_media_out_atmos" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT">
+ <profile name="" format="AUDIO_FORMAT_E_AC3_JOC"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_5POINT1"/>
+ </mixPort>
+ <mixPort name="mixport_bus0_media_out_atmos_pcm" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_7POINT1POINT4"/>
+ </mixPort>
+ <mixPort name="mixport_bus1_navigation_out" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bus2_voice_command_out" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bus3_call_ring_out" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bus4_call_out" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bus5_alarm_out" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bus6_notification_out" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bus7_system_sound_out" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bus100_audio_zone_1" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bus200_audio_zone_2" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="primary input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ </mixPort>
+ <mixPort name="mixport_tuner0" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_input_bus_tone_zone_0" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_input_bus_tone_zone_1" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="bus0_media_out" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="bus0_media_out">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3_JOC"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_7POINT1POINT4"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="bus1_navigation_out" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="bus1_navigation_out">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="bus2_voice_command_out" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="bus2_voice_command_out">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="bus3_call_ring_out" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="bus3_call_ring_out">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="bus4_call_out" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="bus4_call_out">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="bus5_alarm_out" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="bus5_alarm_out">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="bus6_notification_out" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="bus6_notification_out">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="bus7_system_sound_out" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="bus7_system_sound_out">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="bus100_audio_zone_1" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="bus100_audio_zone_1">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="bus200_audio_zone_2" role="sink" type="AUDIO_DEVICE_OUT_BUS"
+ address="bus200_audio_zone_2">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ </devicePort>
+ <devicePort tagName="Built-In Back Mic" type="AUDIO_DEVICE_IN_BACK_MIC"
+ role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ </devicePort>
+ <devicePort tagName="Echo-Reference Mic" type="AUDIO_DEVICE_IN_ECHO_REFERENCE"
+ role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ </devicePort>
+ <devicePort tagName="FM Tuner" type="AUDIO_DEVICE_IN_FM_TUNER" role="source"
+ address="tuner0">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="Tone Generator 0" type="AUDIO_DEVICE_IN_BUS" role="source"
+ address="input_bus_tone_zone_0">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="Tone Generator 1" type="AUDIO_DEVICE_IN_BUS" role="source"
+ address="input_bus_tone_zone_1">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ <gains>
+ <gain name="" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-3200" maxValueMB="600" defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ </devicePorts>
+ <!-- route declaration, i.e. list all available sources for a given sink -->
+ <routes>
+ <route type="mix" sink="bus0_media_out"
+ sources="mixport_bus0_media_out,mixport_bus0_media_out_atmos,mixport_bus0_media_out_atmos_pcm"/>
+ <route type="mix" sink="bus1_navigation_out" sources="mixport_bus1_navigation_out"/>
+ <route type="mix" sink="bus2_voice_command_out"
+ sources="mixport_bus2_voice_command_out"/>
+ <route type="mix" sink="bus3_call_ring_out" sources="mixport_bus3_call_ring_out"/>
+ <route type="mix" sink="bus4_call_out" sources="mixport_bus4_call_out"/>
+ <route type="mix" sink="bus5_alarm_out" sources="mixport_bus5_alarm_out"/>
+ <route type="mix" sink="bus6_notification_out"
+ sources="mixport_bus6_notification_out"/>
+ <route type="mix" sink="bus7_system_sound_out"
+ sources="mixport_bus7_system_sound_out"/>
+ <route type="mix" sink="bus100_audio_zone_1" sources="mixport_bus100_audio_zone_1"/>
+ <route type="mix" sink="bus200_audio_zone_2" sources="mixport_bus200_audio_zone_2"/>
+ <route type="mix" sink="primary input"
+ sources="Built-In Mic,Built-In Back Mic,Echo-Reference Mic"/>
+ <route type="mix" sink="mixport_tuner0" sources="FM Tuner"/>
+ <route type="mix" sink="mixport_input_bus_tone_zone_0" sources="Tone Generator 0"/>
+ <route type="mix" sink="mixport_input_bus_tone_zone_1" sources="Tone Generator 1"/>
+ </routes>
+ </module>
+ </modules>
+</audioPolicyConfiguration>
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index dec439f..7c2f34f 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -2291,6 +2291,9 @@
mOperatingMode = operatingMode;
}
+ // Reset min expected duration when session is reconfigured.
+ mMinExpectedDuration = 0;
+
// In case called from configureStreams, abort queued input buffers not belonging to
// any pending requests.
if (mInputStream != NULL && notifyRequestThread) {
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
index 8e4ff13..69163a5 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
@@ -18,6 +18,7 @@
#define ATRACE_TAG ATRACE_TAG_CAMERA
//#define LOG_NDEBUG 0
+#include <algorithm>
#include <ctime>
#include <fstream>
@@ -1402,14 +1403,30 @@
const VsyncEventData& vsyncEventData = parcelableVsyncEventData.vsync;
nsecs_t currentTime = systemTime();
- // Reset capture to present time offset if more than 1 second
- // between frames.
- if (t - mLastCaptureTime > kSpacingResetIntervalNs) {
+ // Reset capture to present time offset if:
+ // - More than 1 second between frames.
+ // - The frame duration deviates from multiples of vsync frame intervals.
+ nsecs_t captureInterval = t - mLastCaptureTime;
+ float captureToVsyncIntervalRatio = 1.0f * captureInterval / vsyncEventData.frameInterval;
+ float ratioDeviation = std::fabs(
+ captureToVsyncIntervalRatio - std::roundf(captureToVsyncIntervalRatio));
+ if (captureInterval > kSpacingResetIntervalNs ||
+ ratioDeviation >= kMaxIntervalRatioDeviation) {
+ nsecs_t minPresentT = mLastPresentTime + vsyncEventData.frameInterval / 2;
for (size_t i = 0; i < VsyncEventData::kFrameTimelinesLength; i++) {
- if (vsyncEventData.frameTimelines[i].deadlineTimestamp >= currentTime) {
- mCaptureToPresentOffset =
- vsyncEventData.frameTimelines[i].expectedPresentationTime - t;
- break;
+ const auto& timeline = vsyncEventData.frameTimelines[i];
+ if (timeline.deadlineTimestamp >= currentTime &&
+ timeline.expectedPresentationTime > minPresentT) {
+ nsecs_t presentT = vsyncEventData.frameTimelines[i].expectedPresentationTime;
+ mCaptureToPresentOffset = presentT - t;
+ mLastCaptureTime = t;
+ mLastPresentTime = presentT;
+
+ // Move the expected presentation time back by 1/3 of frame interval to
+ // mitigate the time drift. Due to time drift, if we directly use the
+ // expected presentation time, often times 2 expected presentation time
+ // falls into the same VSYNC interval.
+ return presentT - vsyncEventData.frameInterval/3;
}
}
}
@@ -1425,16 +1442,27 @@
int minVsyncs = (mMinExpectedDuration - vsyncEventData.frameInterval / 2) /
vsyncEventData.frameInterval;
if (minVsyncs < 0) minVsyncs = 0;
- nsecs_t minInterval = minVsyncs * vsyncEventData.frameInterval + kTimelineThresholdNs;
- // Find best timestamp in the vsync timeline:
+ nsecs_t minInterval = minVsyncs * vsyncEventData.frameInterval;
+ // Find best timestamp in the vsync timelines:
+ // - Only use at most 3 timelines to avoid long latency
// - closest to the ideal present time,
// - deadline timestamp is greater than the current time, and
// - the candidate present time is at least minInterval in the future
// compared to last present time.
- for (const auto& vsyncTime : vsyncEventData.frameTimelines) {
+ int maxTimelines = std::min(kMaxTimelines, (int)VsyncEventData::kFrameTimelinesLength);
+ float biasForShortDelay = 1.0f;
+ for (int i = 0; i < maxTimelines; i ++) {
+ const auto& vsyncTime = vsyncEventData.frameTimelines[i];
+ if (minVsyncs > 0) {
+ // Bias towards using smaller timeline index:
+ // i = 0: bias = 1
+ // i = maxTimelines-1: bias = -1
+ biasForShortDelay = 1.0 - 2.0 * i / (maxTimelines - 1);
+ }
if (std::abs(vsyncTime.expectedPresentationTime - idealPresentT) < minDiff &&
vsyncTime.deadlineTimestamp >= currentTime &&
- vsyncTime.expectedPresentationTime > mLastPresentTime + minInterval) {
+ vsyncTime.expectedPresentationTime >
+ mLastPresentTime + minInterval + biasForShortDelay * kTimelineThresholdNs) {
expectedPresentT = vsyncTime.expectedPresentationTime;
minDiff = std::abs(vsyncTime.expectedPresentationTime - idealPresentT);
}
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.h b/services/camera/libcameraservice/device3/Camera3OutputStream.h
index 4ab052b..3587af4 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.h
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.h
@@ -426,8 +426,10 @@
nsecs_t mLastPresentTime = 0;
nsecs_t mCaptureToPresentOffset = 0;
static constexpr size_t kDisplaySyncExtraBuffer = 2;
- static constexpr nsecs_t kSpacingResetIntervalNs = 1000000000LL; // 1 second
+ static constexpr nsecs_t kSpacingResetIntervalNs = 50000000LL; // 50 millisecond
static constexpr nsecs_t kTimelineThresholdNs = 1000000LL; // 1 millisecond
+ static constexpr float kMaxIntervalRatioDeviation = 0.05f;
+ static constexpr int kMaxTimelines = 3;
nsecs_t syncTimestampToDisplayLocked(nsecs_t t);
// Re-space frames by delaying queueBuffer so that frame delivery has
diff --git a/services/camera/libcameraservice/utils/TagMonitor.cpp b/services/camera/libcameraservice/utils/TagMonitor.cpp
index 461f5e9..fe87ed6 100644
--- a/services/camera/libcameraservice/utils/TagMonitor.cpp
+++ b/services/camera/libcameraservice/utils/TagMonitor.cpp
@@ -169,6 +169,22 @@
camera_metadata_entry lastEntry = lastValues.find(tag);
+ // Monitor when the stream ids change, this helps visually see what
+ // monitored metadata values are for capture requests with different
+ // stream ids.
+ if (source == REQUEST) {
+ if (inputStreamId != mLastInputStreamId) {
+ mMonitoringEvents.emplace(source, frameNumber, timestamp, camera_metadata_ro_entry_t{},
+ cameraId, std::unordered_set<int>(), inputStreamId);
+ mLastInputStreamId = inputStreamId;
+ }
+
+ if (outputStreamIds != mLastStreamIds) {
+ mMonitoringEvents.emplace(source, frameNumber, timestamp, camera_metadata_ro_entry_t{},
+ cameraId, outputStreamIds, -1);
+ mLastStreamIds = outputStreamIds;
+ }
+ }
if (entry.count > 0) {
bool isDifferent = false;
if (lastEntry.count > 0) {
@@ -190,22 +206,14 @@
// No last entry, so always consider to be different
isDifferent = true;
}
- // Also monitor when the stream ids change, this helps visually see what
- // monitored metadata values are for capture requests with different
- // stream ids.
- if (source == REQUEST &&
- (inputStreamId != mLastInputStreamId || outputStreamIds != mLastStreamIds)) {
- mLastInputStreamId = inputStreamId;
- mLastStreamIds = outputStreamIds;
- isDifferent = true;
- }
+
if (isDifferent) {
ALOGV("%s: Tag %s changed", __FUNCTION__,
get_local_camera_metadata_tag_name_vendor_id(
tag, mVendorTagId));
lastValues.update(entry);
mMonitoringEvents.emplace(source, frameNumber, timestamp, entry, cameraId,
- outputStreamIds, inputStreamId);
+ std::unordered_set<int>(), -1);
}
} else if (lastEntry.count > 0) {
// Value has been removed
@@ -219,8 +227,8 @@
entry.count = 0;
mLastInputStreamId = inputStreamId;
mLastStreamIds = outputStreamIds;
- mMonitoringEvents.emplace(source, frameNumber, timestamp, entry, cameraId, outputStreamIds,
- inputStreamId);
+ mMonitoringEvents.emplace(source, frameNumber, timestamp, entry, cameraId,
+ std::unordered_set<int>(), -1);
}
}
@@ -261,23 +269,39 @@
for (const auto& event : mMonitoringEvents) {
int indentation = (event.source == REQUEST) ? 15 : 30;
- String8 eventString = String8::format("f%d:%" PRId64 "ns:%*s%*s%s.%s: ",
+ String8 eventString = String8::format("f%d:%" PRId64 "ns:%*s%*s",
event.frameNumber, event.timestamp,
2, event.cameraId.c_str(),
indentation,
- event.source == REQUEST ? "REQ:" : "RES:",
+ event.source == REQUEST ? "REQ:" : "RES:");
+
+ if (!event.outputStreamIds.empty()) {
+ eventString += " output stream ids:";
+ for (const auto& id : event.outputStreamIds) {
+ eventString.appendFormat(" %d", id);
+ }
+ eventString += "\n";
+ vec.emplace_back(eventString.string());
+ continue;
+ }
+
+ if (event.inputStreamId != -1) {
+ eventString.appendFormat(" input stream id: %d\n", event.inputStreamId);
+ vec.emplace_back(eventString.string());
+ continue;
+ }
+
+ eventString += String8::format(
+ "%s.%s: ",
get_local_camera_metadata_section_name_vendor_id(event.tag, mVendorTagId),
get_local_camera_metadata_tag_name_vendor_id(event.tag, mVendorTagId));
- if (event.newData.size() == 0) {
- eventString += " (Removed)";
+
+ if (event.newData.empty()) {
+ eventString += " (Removed)\n";
} else {
- eventString += getEventDataString(event.newData.data(),
- event.tag,
- event.type,
- event.newData.size() / camera_metadata_type_size[event.type],
- indentation + 18,
- event.outputStreamIds,
- event.inputStreamId);
+ eventString += getEventDataString(
+ event.newData.data(), event.tag, event.type,
+ event.newData.size() / camera_metadata_type_size[event.type], indentation + 18);
}
vec.emplace_back(eventString.string());
}
@@ -285,13 +309,8 @@
#define CAMERA_METADATA_ENUM_STRING_MAX_SIZE 29
-String8 TagMonitor::getEventDataString(const uint8_t* data_ptr,
- uint32_t tag,
- int type,
- int count,
- int indentation,
- const std::unordered_set<int32_t>& outputStreamIds,
- int32_t inputStreamId) {
+String8 TagMonitor::getEventDataString(const uint8_t* data_ptr, uint32_t tag, int type, int count,
+ int indentation) {
static int values_per_line[NUM_TYPES] = {
[TYPE_BYTE] = 16,
[TYPE_INT32] = 8,
@@ -362,17 +381,7 @@
returnStr += "??? ";
}
}
- returnStr += "] ";
- if (!outputStreamIds.empty()) {
- returnStr += "output stream ids: ";
- for (const auto &id : outputStreamIds) {
- returnStr.appendFormat(" %d ", id);
- }
- }
- if (inputStreamId != -1) {
- returnStr.appendFormat("input stream id: %d", inputStreamId);
- }
- returnStr += "\n";
+ returnStr += "]\n";
}
return returnStr;
}
@@ -385,11 +394,12 @@
source(src),
frameNumber(frameNumber),
timestamp(timestamp),
+ cameraId(cameraId),
tag(value.tag),
type(value.type),
newData(value.data.u8, value.data.u8 + camera_metadata_type_size[value.type] * value.count),
- cameraId(cameraId), outputStreamIds(outputStreamIds), inputStreamId(inputStreamId) {
-}
+ outputStreamIds(outputStreamIds),
+ inputStreamId(inputStreamId) {}
TagMonitor::MonitorEvent::~MonitorEvent() {
}
diff --git a/services/camera/libcameraservice/utils/TagMonitor.h b/services/camera/libcameraservice/utils/TagMonitor.h
index 088d6fe..9ded15d 100644
--- a/services/camera/libcameraservice/utils/TagMonitor.h
+++ b/services/camera/libcameraservice/utils/TagMonitor.h
@@ -85,12 +85,8 @@
// function.
void dumpMonitoredTagEventsToVectorLocked(std::vector<std::string> &out);
- static String8 getEventDataString(const uint8_t *data_ptr,
- uint32_t tag, int type,
- int count,
- int indentation,
- const std::unordered_set<int32_t> &outputStreamIds,
- int32_t inputStreamId);
+ static String8 getEventDataString(const uint8_t* data_ptr, uint32_t tag, int type, int count,
+ int indentation);
void monitorSingleMetadata(TagMonitor::eventSource source, int64_t frameNumber,
nsecs_t timestamp, const std::string& cameraId, uint32_t tag,
@@ -128,12 +124,15 @@
eventSource source;
uint32_t frameNumber;
nsecs_t timestamp;
+ std::string cameraId;
uint32_t tag;
uint8_t type;
std::vector<uint8_t> newData;
- std::string cameraId;
+ // NOTE: We want to print changes to outputStreamIds and inputStreamId in their own lines.
+ // So any MonitorEvent where these fields are not the default value will have garbage
+ // values for all fields other than source, frameNumber, timestamp, and cameraId.
std::unordered_set<int32_t> outputStreamIds;
- int32_t inputStreamId = 1;
+ int32_t inputStreamId = -1;
};
// A ring buffer for tracking the last kMaxMonitorEvents metadata changes