Merge "C2AllocatorBlob: allow multiple maps"
diff --git a/OWNERS b/OWNERS
index 8f405e9..9989bf0 100644
--- a/OWNERS
+++ b/OWNERS
@@ -2,3 +2,7 @@
etalvala@google.com
lajos@google.com
marcone@google.com
+
+# LON
+olly@google.com
+andrewlewis@google.com
diff --git a/cmds/stagefright/AudioPlayer.cpp b/cmds/stagefright/AudioPlayer.cpp
index eb76953..55427ca 100644
--- a/cmds/stagefright/AudioPlayer.cpp
+++ b/cmds/stagefright/AudioPlayer.cpp
@@ -134,15 +134,18 @@
success = format->findInt32(kKeySampleRate, &mSampleRate);
CHECK(success);
- int32_t numChannels, channelMask;
+ int32_t numChannels;
success = format->findInt32(kKeyChannelCount, &numChannels);
CHECK(success);
- if(!format->findInt32(kKeyChannelMask, &channelMask)) {
+ audio_channel_mask_t channelMask;
+ if (int32_t rawChannelMask; !format->findInt32(kKeyChannelMask, &rawChannelMask)) {
// log only when there's a risk of ambiguity of channel mask selection
ALOGI_IF(numChannels > 2,
"source format didn't specify channel mask, using (%d) channel order", numChannels);
channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
+ } else {
+ channelMask = static_cast<audio_channel_mask_t>(rawChannelMask);
}
audio_format_t audioFormat = AUDIO_FORMAT_PCM_16_BIT;
diff --git a/include/media/MicrophoneInfo.h b/include/media/MicrophoneInfo.h
index 2287aca..0a24b02 100644
--- a/include/media/MicrophoneInfo.h
+++ b/include/media/MicrophoneInfo.h
@@ -205,11 +205,11 @@
private:
status_t readFloatVector(
const Parcel* parcel, Vector<float> *vectorPtr, size_t defaultLength) {
- std::unique_ptr<std::vector<float>> v;
+ std::optional<std::vector<float>> v;
status_t result = parcel->readFloatVector(&v);
if (result != OK) return result;
vectorPtr->clear();
- if (v.get() != nullptr) {
+ if (v) {
for (const auto& iter : *v) {
vectorPtr->push_back(iter);
}
diff --git a/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoEncTest.cpp b/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoEncTest.cpp
index ecaf3a8..5bcea5b 100644
--- a/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoEncTest.cpp
+++ b/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoEncTest.cpp
@@ -107,6 +107,13 @@
mOutputSize = 0u;
mTimestampDevTest = false;
if (mCompName == unknown_comp) mDisableTest = true;
+
+ C2SecureModeTuning secureModeTuning{};
+ mComponent->query({&secureModeTuning}, {}, C2_MAY_BLOCK, nullptr);
+ if (secureModeTuning.value == C2Config::SM_READ_PROTECTED) {
+ mDisableTest = true;
+ }
+
if (mDisableTest) std::cout << "[ WARN ] Test Disabled \n";
}
diff --git a/media/codec2/hidl/client/client.cpp b/media/codec2/hidl/client/client.cpp
index 7e4352d..4650672 100644
--- a/media/codec2/hidl/client/client.cpp
+++ b/media/codec2/hidl/client/client.cpp
@@ -843,6 +843,11 @@
return;
}
});
+ if (!transStatus.isOk()) {
+ LOG(DEBUG) << "SimpleParamReflector -- transaction failed: "
+ << transStatus.description();
+ descriptor.reset();
+ }
return descriptor;
}
diff --git a/media/codec2/sfplugin/C2OMXNode.cpp b/media/codec2/sfplugin/C2OMXNode.cpp
index c7588e9..dd1f485 100644
--- a/media/codec2/sfplugin/C2OMXNode.cpp
+++ b/media/codec2/sfplugin/C2OMXNode.cpp
@@ -25,6 +25,7 @@
#include <C2AllocatorGralloc.h>
#include <C2BlockInternal.h>
#include <C2Component.h>
+#include <C2Config.h>
#include <C2PlatformSupport.h>
#include <OMX_Component.h>
@@ -44,6 +45,8 @@
namespace {
+constexpr OMX_U32 kPortIndexInput = 0;
+
class Buffer2D : public C2Buffer {
public:
explicit Buffer2D(C2ConstGraphicBlock block) : C2Buffer({ block }) {}
@@ -200,11 +203,27 @@
return BAD_VALUE;
}
OMX_PARAM_PORTDEFINITIONTYPE *pDef = (OMX_PARAM_PORTDEFINITIONTYPE *)params;
- // TODO: read these from intf()
+ if (pDef->nPortIndex != kPortIndexInput) {
+ break;
+ }
+
pDef->nBufferCountActual = 16;
+
+ std::shared_ptr<Codec2Client::Component> comp = mComp.lock();
+ C2PortActualDelayTuning::input inputDelay(0);
+ C2ActualPipelineDelayTuning pipelineDelay(0);
+ c2_status_t c2err = comp->query(
+ {&inputDelay, &pipelineDelay}, {}, C2_DONT_BLOCK, nullptr);
+ if (c2err == C2_OK || c2err == C2_BAD_INDEX) {
+ pDef->nBufferCountActual = 4;
+ pDef->nBufferCountActual += (inputDelay ? inputDelay.value : 0u);
+ pDef->nBufferCountActual += (pipelineDelay ? pipelineDelay.value : 0u);
+ }
+
pDef->eDomain = OMX_PortDomainVideo;
pDef->format.video.nFrameWidth = mWidth;
pDef->format.video.nFrameHeight = mHeight;
+ pDef->format.video.eColorFormat = OMX_COLOR_FormatAndroidOpaque;
err = OK;
break;
}
diff --git a/media/codec2/sfplugin/CCodec.cpp b/media/codec2/sfplugin/CCodec.cpp
index 1405b97..57252b2 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -246,8 +246,19 @@
if (source == nullptr) {
return NO_INIT;
}
- constexpr size_t kNumSlots = 16;
- for (size_t i = 0; i < kNumSlots; ++i) {
+
+ size_t numSlots = 4;
+ constexpr OMX_U32 kPortIndexInput = 0;
+
+ OMX_PARAM_PORTDEFINITIONTYPE param;
+ param.nPortIndex = kPortIndexInput;
+ status_t err = mNode->getParameter(OMX_IndexParamPortDefinition,
+ ¶m, sizeof(param));
+ if (err == OK) {
+ numSlots = param.nBufferCountActual;
+ }
+
+ for (size_t i = 0; i < numSlots; ++i) {
source->onInputBufferAdded(i);
}
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 3919ea2..aa4a004 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -1066,9 +1066,6 @@
Mutexed<OutputSurface>::Locked output(mOutputSurface);
output->maxDequeueBuffers = numOutputSlots +
reorderDepth.value + kRenderingDepth;
- if (!secure) {
- output->maxDequeueBuffers += numInputSlots;
- }
outputSurface = output->surface ?
output->surface->getIGraphicBufferProducer() : nullptr;
if (outputSurface) {
@@ -1529,6 +1526,7 @@
}
std::optional<uint32_t> newInputDelay, newPipelineDelay;
+ bool needMaxDequeueBufferCountUpdate = false;
while (!worklet->output.configUpdate.empty()) {
std::unique_ptr<C2Param> param;
worklet->output.configUpdate.back().swap(param);
@@ -1537,24 +1535,10 @@
case C2PortReorderBufferDepthTuning::CORE_INDEX: {
C2PortReorderBufferDepthTuning::output reorderDepth;
if (reorderDepth.updateFrom(*param)) {
- bool secure = mComponent->getName().find(".secure") !=
- std::string::npos;
- mOutput.lock()->buffers->setReorderDepth(
- reorderDepth.value);
ALOGV("[%s] onWorkDone: updated reorder depth to %u",
mName, reorderDepth.value);
- size_t numOutputSlots = mOutput.lock()->numSlots;
- size_t numInputSlots = mInput.lock()->numSlots;
- Mutexed<OutputSurface>::Locked output(mOutputSurface);
- output->maxDequeueBuffers = numOutputSlots +
- reorderDepth.value + kRenderingDepth;
- if (!secure) {
- output->maxDequeueBuffers += numInputSlots;
- }
- if (output->surface) {
- output->surface->setMaxDequeuedBufferCount(
- output->maxDequeueBuffers);
- }
+ mOutput.lock()->buffers->setReorderDepth(reorderDepth.value);
+ needMaxDequeueBufferCountUpdate = true;
} else {
ALOGD("[%s] onWorkDone: failed to read reorder depth",
mName);
@@ -1598,14 +1582,11 @@
if (outputDelay.updateFrom(*param)) {
ALOGV("[%s] onWorkDone: updating output delay %u",
mName, outputDelay.value);
- bool secure = mComponent->getName().find(".secure") !=
- std::string::npos;
- (void)mPipelineWatcher.lock()->outputDelay(
- outputDelay.value);
+ (void)mPipelineWatcher.lock()->outputDelay(outputDelay.value);
+ needMaxDequeueBufferCountUpdate = true;
bool outputBuffersChanged = false;
size_t numOutputSlots = 0;
- size_t numInputSlots = mInput.lock()->numSlots;
{
Mutexed<Output>::Locked output(mOutput);
if (!output->buffers) {
@@ -1631,16 +1612,6 @@
if (outputBuffersChanged) {
mCCodecCallback->onOutputBuffersChanged();
}
-
- uint32_t depth = mOutput.lock()->buffers->getReorderDepth();
- Mutexed<OutputSurface>::Locked output(mOutputSurface);
- output->maxDequeueBuffers = numOutputSlots + depth + kRenderingDepth;
- if (!secure) {
- output->maxDequeueBuffers += numInputSlots;
- }
- if (output->surface) {
- output->surface->setMaxDequeuedBufferCount(output->maxDequeueBuffers);
- }
}
}
break;
@@ -1669,6 +1640,20 @@
input->numSlots = newNumSlots;
}
}
+ if (needMaxDequeueBufferCountUpdate) {
+ size_t numOutputSlots = 0;
+ uint32_t reorderDepth = 0;
+ {
+ Mutexed<Output>::Locked output(mOutput);
+ numOutputSlots = output->numSlots;
+ reorderDepth = output->buffers->getReorderDepth();
+ }
+ Mutexed<OutputSurface>::Locked output(mOutputSurface);
+ output->maxDequeueBuffers = numOutputSlots + reorderDepth + kRenderingDepth;
+ if (output->surface) {
+ output->surface->setMaxDequeuedBufferCount(output->maxDequeueBuffers);
+ }
+ }
int32_t flags = 0;
if (worklet->output.flags & C2FrameData::FLAG_END_OF_STREAM) {
diff --git a/media/codec2/sfplugin/CCodecConfig.cpp b/media/codec2/sfplugin/CCodecConfig.cpp
index 96f86e8..79c6227 100644
--- a/media/codec2/sfplugin/CCodecConfig.cpp
+++ b/media/codec2/sfplugin/CCodecConfig.cpp
@@ -1151,14 +1151,11 @@
bool changed = false;
if (domain & mInputDomain) {
- sp<AMessage> oldFormat = mInputFormat;
- mInputFormat = mInputFormat->dup(); // trigger format changed
+ sp<AMessage> oldFormat = mInputFormat->dup();
mInputFormat->extend(getFormatForDomain(reflected, mInputDomain));
if (mInputFormat->countEntries() != oldFormat->countEntries()
|| mInputFormat->changesFrom(oldFormat)->countEntries() > 0) {
changed = true;
- } else {
- mInputFormat = oldFormat; // no change
}
}
if (domain & mOutputDomain) {
diff --git a/media/extractors/ogg/OggExtractor.cpp b/media/extractors/ogg/OggExtractor.cpp
index 828bcd6..eb2246d 100644
--- a/media/extractors/ogg/OggExtractor.cpp
+++ b/media/extractors/ogg/OggExtractor.cpp
@@ -1304,8 +1304,8 @@
|| audioChannelCount <= 0 || audioChannelCount > FCC_8) {
ALOGE("Invalid haptic channel count found in metadata: %d", mHapticChannelCount);
} else {
- const audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(
- audioChannelCount) | hapticChannelMask;
+ const audio_channel_mask_t channelMask = static_cast<audio_channel_mask_t>(
+ audio_channel_out_mask_from_count(audioChannelCount) | hapticChannelMask);
AMediaFormat_setInt32(mMeta, AMEDIAFORMAT_KEY_CHANNEL_MASK, channelMask);
AMediaFormat_setInt32(
mMeta, AMEDIAFORMAT_KEY_HAPTIC_CHANNEL_COUNT, mHapticChannelCount);
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index 9007b10..dbb3d2b 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -231,7 +231,8 @@
case AAUDIO_ALLOW_CAPTURE_BY_SYSTEM:
return AUDIO_FLAG_NO_MEDIA_PROJECTION;
case AAUDIO_ALLOW_CAPTURE_BY_NONE:
- return AUDIO_FLAG_NO_MEDIA_PROJECTION | AUDIO_FLAG_NO_SYSTEM_CAPTURE;
+ return static_cast<audio_flags_mask_t>(
+ AUDIO_FLAG_NO_MEDIA_PROJECTION | AUDIO_FLAG_NO_SYSTEM_CAPTURE);
default:
ALOGE("%s() 0x%08X unrecognized", __func__, policy);
return AUDIO_FLAG_NONE; //
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 509e063..d97f6cf 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -279,7 +279,8 @@
mAttributes.source = inputSource;
if (inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION
|| inputSource == AUDIO_SOURCE_CAMCORDER) {
- mAttributes.flags |= AUDIO_FLAG_CAPTURE_PRIVATE;
+ mAttributes.flags = static_cast<audio_flags_mask_t>(
+ mAttributes.flags | AUDIO_FLAG_CAPTURE_PRIVATE);
}
} else {
// stream type shouldn't be looked at, this track has audio attributes
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 807aa13..721873a 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -222,7 +222,7 @@
{
mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
mAttributes.usage = AUDIO_USAGE_UNKNOWN;
- mAttributes.flags = 0x0;
+ mAttributes.flags = AUDIO_FLAG_NONE;
strcpy(mAttributes.tags, "");
}
@@ -458,7 +458,7 @@
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
} else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
- mAttributes.flags |= AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO;
+ flags = static_cast<audio_output_flags_t>(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
}
// validate parameters
@@ -635,6 +635,36 @@
return status;
}
+
+status_t AudioTrack::set(
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ size_t frameCount,
+ audio_output_flags_t flags,
+ callback_t cbf,
+ void* user,
+ int32_t notificationFrames,
+ const sp<IMemory>& sharedBuffer,
+ bool threadCanCallJava,
+ audio_session_t sessionId,
+ transfer_type transferType,
+ const audio_offload_info_t *offloadInfo,
+ uid_t uid,
+ pid_t pid,
+ const audio_attributes_t* pAttributes,
+ bool doNotReconnect,
+ float maxRequiredSpeed,
+ audio_port_handle_t selectedDeviceId)
+{
+ return set(streamType, sampleRate, format,
+ static_cast<audio_channel_mask_t>(channelMask),
+ frameCount, flags, cbf, user, notificationFrames, sharedBuffer,
+ threadCanCallJava, sessionId, transferType, offloadInfo, uid, pid,
+ pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
+}
+
// -------------------------------------------------------------------------
status_t AudioTrack::start()
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 6d79aba..04525d0 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -1210,7 +1210,7 @@
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
- audio_channel_mask_t channelMask = data.readInt32();
+ audio_channel_mask_t channelMask = (audio_channel_mask_t) data.readInt32();
reply->writeInt64( getInputBufferSize(sampleRate, format, channelMask) );
return NO_ERROR;
} break;
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index 60af84b..c9bb0aa 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -2628,7 +2628,7 @@
case SET_ALLOWED_CAPTURE_POLICY: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
uid_t uid = data.readInt32();
- audio_flags_mask_t flags = data.readInt32();
+ audio_flags_mask_t flags = static_cast<audio_flags_mask_t>(data.readInt32());
status_t status = setAllowedCapturePolicy(uid, flags);
reply->writeInt32(status);
return NO_ERROR;
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index 0dbd842..ca51dcd 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -338,6 +338,27 @@
bool doNotReconnect = false,
float maxRequiredSpeed = 1.0f,
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
+ // FIXME(b/169889714): Vendor code depends on the old method signature at link time
+ status_t set(audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ size_t frameCount = 0,
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ callback_t cbf = NULL,
+ void* user = NULL,
+ int32_t notificationFrames = 0,
+ const sp<IMemory>& sharedBuffer = 0,
+ bool threadCanCallJava = false,
+ audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
+ transfer_type transferType = TRANSFER_DEFAULT,
+ const audio_offload_info_t *offloadInfo = NULL,
+ uid_t uid = AUDIO_UID_INVALID,
+ pid_t pid = -1,
+ const audio_attributes_t* pAttributes = NULL,
+ bool doNotReconnect = false,
+ float maxRequiredSpeed = 1.0f,
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
/* Result of constructing the AudioTrack. This must be checked for successful initialization
* before using any AudioTrack API (except for set()), because using
diff --git a/media/libaudiofoundation/AudioDeviceTypeAddr.cpp b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
index b44043a..d65d6ac 100644
--- a/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
+++ b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
@@ -43,7 +43,9 @@
status_t AudioDeviceTypeAddr::readFromParcel(const Parcel *parcel) {
status_t status;
- if ((status = parcel->readUint32(&mType)) != NO_ERROR) return status;
+ uint32_t rawDeviceType;
+ if ((status = parcel->readUint32(&rawDeviceType)) != NO_ERROR) return status;
+ mType = static_cast<audio_devices_t>(rawDeviceType);
status = parcel->readUtf8FromUtf16(&mAddress);
return status;
}
@@ -64,4 +66,4 @@
return deviceTypes;
}
-}
\ No newline at end of file
+}
diff --git a/media/libaudiofoundation/AudioGain.cpp b/media/libaudiofoundation/AudioGain.cpp
index 0d28335..759140e 100644
--- a/media/libaudiofoundation/AudioGain.cpp
+++ b/media/libaudiofoundation/AudioGain.cpp
@@ -152,8 +152,12 @@
if ((status = parcel->readInt32(&mIndex)) != NO_ERROR) return status;
if ((status = parcel->readBool(&mUseInChannelMask)) != NO_ERROR) return status;
if ((status = parcel->readBool(&mUseForVolume)) != NO_ERROR) return status;
- if ((status = parcel->readUint32(&mGain.mode)) != NO_ERROR) return status;
- if ((status = parcel->readUint32(&mGain.channel_mask)) != NO_ERROR) return status;
+ uint32_t rawGainMode;
+ if ((status = parcel->readUint32(&rawGainMode)) != NO_ERROR) return status;
+ mGain.mode = static_cast<audio_gain_mode_t>(rawGainMode);
+ uint32_t rawChannelMask;
+ if ((status = parcel->readUint32(&rawChannelMask)) != NO_ERROR) return status;
+ mGain.channel_mask = static_cast<audio_channel_mask_t>(rawChannelMask);
if ((status = parcel->readInt32(&mGain.min_value)) != NO_ERROR) return status;
if ((status = parcel->readInt32(&mGain.max_value)) != NO_ERROR) return status;
if ((status = parcel->readInt32(&mGain.default_value)) != NO_ERROR) return status;
diff --git a/media/libaudiofoundation/AudioPort.cpp b/media/libaudiofoundation/AudioPort.cpp
index f988690..1846a6b 100644
--- a/media/libaudiofoundation/AudioPort.cpp
+++ b/media/libaudiofoundation/AudioPort.cpp
@@ -268,12 +268,17 @@
if ((status = parcel->readUint32(reinterpret_cast<uint32_t*>(&mFormat))) != NO_ERROR) {
return status;
}
- if ((status = parcel->readUint32(&mChannelMask)) != NO_ERROR) return status;
+ uint32_t rawChannelMask;
+ if ((status = parcel->readUint32(&rawChannelMask)) != NO_ERROR) return status;
+ mChannelMask = static_cast<audio_channel_mask_t>(rawChannelMask);
if ((status = parcel->readInt32(&mId)) != NO_ERROR) return status;
// Read mGain from parcel.
if ((status = parcel->readInt32(&mGain.index)) != NO_ERROR) return status;
- if ((status = parcel->readUint32(&mGain.mode)) != NO_ERROR) return status;
- if ((status = parcel->readUint32(&mGain.channel_mask)) != NO_ERROR) return status;
+ uint32_t rawGainMode;
+ if ((status = parcel->readUint32(&rawGainMode)) != NO_ERROR) return status;
+ mGain.mode = static_cast<audio_gain_mode_t>(rawGainMode);
+ if ((status = parcel->readUint32(&rawChannelMask)) != NO_ERROR) return status;
+ mGain.channel_mask = static_cast<audio_channel_mask_t>(rawChannelMask);
if ((status = parcel->readUint32(&mGain.ramp_duration_ms)) != NO_ERROR) return status;
std::vector<int> values;
if ((status = parcel->readInt32Vector(&values)) != NO_ERROR) return status;
diff --git a/media/libaudiofoundation/AudioProfile.cpp b/media/libaudiofoundation/AudioProfile.cpp
index 91be346..67b600e 100644
--- a/media/libaudiofoundation/AudioProfile.cpp
+++ b/media/libaudiofoundation/AudioProfile.cpp
@@ -157,7 +157,9 @@
std::vector<int> values;
if ((status = parcel->readInt32Vector(&values)) != NO_ERROR) return status;
mChannelMasks.clear();
- mChannelMasks.insert(values.begin(), values.end());
+ for (auto raw : values) {
+ mChannelMasks.insert(static_cast<audio_channel_mask_t>(raw));
+ }
values.clear();
if ((status = parcel->readInt32Vector(&values)) != NO_ERROR) return status;
mSamplingRates.clear();
diff --git a/media/libaudiofoundation/include/media/AudioContainers.h b/media/libaudiofoundation/include/media/AudioContainers.h
index 72fda49..aa7ca69 100644
--- a/media/libaudiofoundation/include/media/AudioContainers.h
+++ b/media/libaudiofoundation/include/media/AudioContainers.h
@@ -96,7 +96,7 @@
static inline audio_devices_t deviceTypesToBitMask(const DeviceTypeSet& deviceTypes) {
audio_devices_t types = AUDIO_DEVICE_NONE;
for (auto deviceType : deviceTypes) {
- types |= deviceType;
+ types = static_cast<audio_devices_t>(types | deviceType);
}
return types;
}
@@ -131,4 +131,4 @@
std::string toString(const DeviceTypeSet& deviceTypes);
-} // namespace android
\ No newline at end of file
+} // namespace android
diff --git a/media/libaudiohal/Android.bp b/media/libaudiohal/Android.bp
index 1709d1e..fab0fea 100644
--- a/media/libaudiohal/Android.bp
+++ b/media/libaudiohal/Android.bp
@@ -18,6 +18,7 @@
"libaudiohal@4.0",
"libaudiohal@5.0",
"libaudiohal@6.0",
+// "libaudiohal@7.0",
],
shared_libs: [
diff --git a/media/libaudiohal/FactoryHalHidl.cpp b/media/libaudiohal/FactoryHalHidl.cpp
index 5985ef0..7228b22 100644
--- a/media/libaudiohal/FactoryHalHidl.cpp
+++ b/media/libaudiohal/FactoryHalHidl.cpp
@@ -31,6 +31,7 @@
/** Supported HAL versions, in order of preference.
*/
const char* sAudioHALVersions[] = {
+ "7.0",
"6.0",
"5.0",
"4.0",
diff --git a/media/libaudiohal/impl/Android.bp b/media/libaudiohal/impl/Android.bp
index 967fba1..df006b5 100644
--- a/media/libaudiohal/impl/Android.bp
+++ b/media/libaudiohal/impl/Android.bp
@@ -116,3 +116,20 @@
]
}
+cc_library_shared {
+ enabled: false,
+ name: "libaudiohal@7.0",
+ defaults: ["libaudiohal_default"],
+ shared_libs: [
+ "android.hardware.audio.common@7.0",
+ "android.hardware.audio.common@7.0-util",
+ "android.hardware.audio.effect@7.0",
+ "android.hardware.audio@7.0",
+ ],
+ cflags: [
+ "-DMAJOR_VERSION=7",
+ "-DMINOR_VERSION=0",
+ "-include common/all-versions/VersionMacro.h",
+ ]
+}
+
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index 1a31420..00baea2 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -79,10 +79,14 @@
&& mixerChannelMask == (track->mMixerChannelMask | track->mMixerHapticChannelMask)) {
return false; // no need to change
}
- const audio_channel_mask_t hapticChannelMask = trackChannelMask & AUDIO_CHANNEL_HAPTIC_ALL;
- trackChannelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
- const audio_channel_mask_t mixerHapticChannelMask = mixerChannelMask & AUDIO_CHANNEL_HAPTIC_ALL;
- mixerChannelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
+ const audio_channel_mask_t hapticChannelMask =
+ static_cast<audio_channel_mask_t>(trackChannelMask & AUDIO_CHANNEL_HAPTIC_ALL);
+ trackChannelMask = static_cast<audio_channel_mask_t>(
+ trackChannelMask & ~AUDIO_CHANNEL_HAPTIC_ALL);
+ const audio_channel_mask_t mixerHapticChannelMask = static_cast<audio_channel_mask_t>(
+ mixerChannelMask & AUDIO_CHANNEL_HAPTIC_ALL);
+ mixerChannelMask = static_cast<audio_channel_mask_t>(
+ mixerChannelMask & ~AUDIO_CHANNEL_HAPTIC_ALL);
// always recompute for both channel masks even if only one has changed.
const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
@@ -362,7 +366,8 @@
const audio_channel_mask_t trackChannelMask =
static_cast<audio_channel_mask_t>(valueInt);
if (setChannelMasks(name, trackChannelMask,
- (track->mMixerChannelMask | track->mMixerHapticChannelMask))) {
+ static_cast<audio_channel_mask_t>(
+ track->mMixerChannelMask | track->mMixerHapticChannelMask))) {
ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
invalidate();
}
@@ -407,7 +412,8 @@
case MIXER_CHANNEL_MASK: {
const audio_channel_mask_t mixerChannelMask =
static_cast<audio_channel_mask_t>(valueInt);
- if (setChannelMasks(name, track->channelMask | track->mHapticChannelMask,
+ if (setChannelMasks(name, static_cast<audio_channel_mask_t>(
+ track->channelMask | track->mHapticChannelMask),
mixerChannelMask)) {
ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
invalidate();
@@ -533,9 +539,10 @@
Track* t = static_cast<Track*>(track);
audio_channel_mask_t channelMask = t->channelMask;
- t->mHapticChannelMask = channelMask & AUDIO_CHANNEL_HAPTIC_ALL;
+ t->mHapticChannelMask = static_cast<audio_channel_mask_t>(
+ channelMask & AUDIO_CHANNEL_HAPTIC_ALL);
t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
- channelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
+ channelMask = static_cast<audio_channel_mask_t>(channelMask & ~AUDIO_CHANNEL_HAPTIC_ALL);
t->channelCount = audio_channel_count_from_out_mask(channelMask);
ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
"Non-stereo channel mask: %d\n", channelMask);
diff --git a/media/libaudioprocessing/tests/test-mixer.cpp b/media/libaudioprocessing/tests/test-mixer.cpp
index bc9d2a6..1bbb863 100644
--- a/media/libaudioprocessing/tests/test-mixer.cpp
+++ b/media/libaudioprocessing/tests/test-mixer.cpp
@@ -241,7 +241,8 @@
// set up the tracks.
for (size_t i = 0; i < providers.size(); ++i) {
//printf("track %d out of %d\n", i, providers.size());
- uint32_t channelMask = audio_channel_out_mask_from_count(providers[i].getNumChannels());
+ audio_channel_mask_t channelMask =
+ audio_channel_out_mask_from_count(providers[i].getNumChannels());
const int name = i;
const status_t status = mixer->create(
name, channelMask, formats[i], AUDIO_SESSION_OUTPUT_MIX);
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 973a164..670b415 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -3394,8 +3394,8 @@
return -EINVAL;
}
- uint32_t device = *(uint32_t*)pCmdData;
- pContext->pBundledContext->nOutputDevice = (audio_devices_t)device;
+ audio_devices_t device = *(audio_devices_t *)pCmdData;
+ pContext->pBundledContext->nOutputDevice = device;
if (pContext->EffectType == LVM_BASS_BOOST) {
if ((device == AUDIO_DEVICE_OUT_SPEAKER) ||
diff --git a/media/libeffects/preprocessing/.clang-format b/media/libeffects/preprocessing/.clang-format
new file mode 120000
index 0000000..f1b4f69
--- /dev/null
+++ b/media/libeffects/preprocessing/.clang-format
@@ -0,0 +1 @@
+../../../../../build/soong/scripts/system-clang-format
\ No newline at end of file
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index f2f74a5..1a5547b 100644
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -18,17 +18,17 @@
#include <string.h>
#define LOG_TAG "PreProcessing"
//#define LOG_NDEBUG 0
-#include <utils/Log.h>
-#include <utils/Timers.h>
-#include <hardware/audio_effect.h>
#include <audio_effects/effect_aec.h>
#include <audio_effects/effect_agc.h>
+#include <hardware/audio_effect.h>
+#include <utils/Log.h>
+#include <utils/Timers.h>
#ifndef WEBRTC_LEGACY
#include <audio_effects/effect_agc2.h>
#endif
#include <audio_effects/effect_ns.h>
-#include <module_common_types.h>
#include <audio_processing.h>
+#include <module_common_types.h>
#ifdef WEBRTC_LEGACY
#include "speex/speex_resampler.h"
#endif
@@ -44,29 +44,28 @@
#define PREPROC_NUM_SESSIONS 8
// types of pre processing modules
-enum preproc_id
-{
- PREPROC_AGC, // Automatic Gain Control
+enum preproc_id {
+ PREPROC_AGC, // Automatic Gain Control
#ifndef WEBRTC_LEGACY
- PREPROC_AGC2, // Automatic Gain Control 2
+ PREPROC_AGC2, // Automatic Gain Control 2
#endif
- PREPROC_AEC, // Acoustic Echo Canceler
- PREPROC_NS, // Noise Suppressor
+ PREPROC_AEC, // Acoustic Echo Canceler
+ PREPROC_NS, // Noise Suppressor
PREPROC_NUM_EFFECTS
};
// Session state
enum preproc_session_state {
- PREPROC_SESSION_STATE_INIT, // initialized
- PREPROC_SESSION_STATE_CONFIG // configuration received
+ PREPROC_SESSION_STATE_INIT, // initialized
+ PREPROC_SESSION_STATE_CONFIG // configuration received
};
// Effect/Preprocessor state
enum preproc_effect_state {
- PREPROC_EFFECT_STATE_INIT, // initialized
- PREPROC_EFFECT_STATE_CREATED, // webRTC engine created
- PREPROC_EFFECT_STATE_CONFIG, // configuration received/disabled
- PREPROC_EFFECT_STATE_ACTIVE // active/enabled
+ PREPROC_EFFECT_STATE_INIT, // initialized
+ PREPROC_EFFECT_STATE_CREATED, // webRTC engine created
+ PREPROC_EFFECT_STATE_CONFIG, // configuration received/disabled
+ PREPROC_EFFECT_STATE_ACTIVE // active/enabled
};
// handle on webRTC engine
@@ -79,95 +78,95 @@
// Effect operation table. Functions for all pre processors are declared in sPreProcOps[] table.
// Function pointer can be null if no action required.
struct preproc_ops_s {
- int (* create)(preproc_effect_t *fx);
- int (* init)(preproc_effect_t *fx);
- int (* reset)(preproc_effect_t *fx);
- void (* enable)(preproc_effect_t *fx);
- void (* disable)(preproc_effect_t *fx);
- int (* set_parameter)(preproc_effect_t *fx, void *param, void *value);
- int (* get_parameter)(preproc_effect_t *fx, void *param, uint32_t *size, void *value);
- int (* set_device)(preproc_effect_t *fx, uint32_t device);
+ int (*create)(preproc_effect_t* fx);
+ int (*init)(preproc_effect_t* fx);
+ int (*reset)(preproc_effect_t* fx);
+ void (*enable)(preproc_effect_t* fx);
+ void (*disable)(preproc_effect_t* fx);
+ int (*set_parameter)(preproc_effect_t* fx, void* param, void* value);
+ int (*get_parameter)(preproc_effect_t* fx, void* param, uint32_t* size, void* value);
+ int (*set_device)(preproc_effect_t* fx, uint32_t device);
};
// Effect context
struct preproc_effect_s {
- const struct effect_interface_s *itfe;
- uint32_t procId; // type of pre processor (enum preproc_id)
- uint32_t state; // current state (enum preproc_effect_state)
- preproc_session_t *session; // session the effect is on
- const preproc_ops_t *ops; // effect ops table
- preproc_fx_handle_t engine; // handle on webRTC engine
- uint32_t type; // subtype of effect
+ const struct effect_interface_s* itfe;
+ uint32_t procId; // type of pre processor (enum preproc_id)
+ uint32_t state; // current state (enum preproc_effect_state)
+ preproc_session_t* session; // session the effect is on
+ const preproc_ops_t* ops; // effect ops table
+ preproc_fx_handle_t engine; // handle on webRTC engine
+ uint32_t type; // subtype of effect
#ifdef DUAL_MIC_TEST
- bool aux_channels_on; // support auxiliary channels
- size_t cur_channel_config; // current auciliary channel configuration
+ bool aux_channels_on; // support auxiliary channels
+ size_t cur_channel_config; // current auciliary channel configuration
#endif
};
// Session context
struct preproc_session_s {
- struct preproc_effect_s effects[PREPROC_NUM_EFFECTS]; // effects in this session
- uint32_t state; // current state (enum preproc_session_state)
- int id; // audio session ID
- int io; // handle of input stream this session is on
- webrtc::AudioProcessing* apm; // handle on webRTC audio processing module (APM)
+ struct preproc_effect_s effects[PREPROC_NUM_EFFECTS]; // effects in this session
+ uint32_t state; // current state (enum preproc_session_state)
+ int id; // audio session ID
+ int io; // handle of input stream this session is on
+ webrtc::AudioProcessing* apm; // handle on webRTC audio processing module (APM)
#ifndef WEBRTC_LEGACY
// Audio Processing module builder
webrtc::AudioProcessingBuilder ap_builder;
#endif
- size_t apmFrameCount; // buffer size for webRTC process (10 ms)
- uint32_t apmSamplingRate; // webRTC APM sampling rate (8/16 or 32 kHz)
- size_t frameCount; // buffer size before input resampler ( <=> apmFrameCount)
- uint32_t samplingRate; // sampling rate at effect process interface
- uint32_t inChannelCount; // input channel count
- uint32_t outChannelCount; // output channel count
- uint32_t createdMsk; // bit field containing IDs of crested pre processors
- uint32_t enabledMsk; // bit field containing IDs of enabled pre processors
- uint32_t processedMsk; // bit field containing IDs of pre processors already
- // processed in current round
+ size_t apmFrameCount; // buffer size for webRTC process (10 ms)
+ uint32_t apmSamplingRate; // webRTC APM sampling rate (8/16 or 32 kHz)
+ size_t frameCount; // buffer size before input resampler ( <=> apmFrameCount)
+ uint32_t samplingRate; // sampling rate at effect process interface
+ uint32_t inChannelCount; // input channel count
+ uint32_t outChannelCount; // output channel count
+ uint32_t createdMsk; // bit field containing IDs of crested pre processors
+ uint32_t enabledMsk; // bit field containing IDs of enabled pre processors
+ uint32_t processedMsk; // bit field containing IDs of pre processors already
+ // processed in current round
#ifdef WEBRTC_LEGACY
- webrtc::AudioFrame *procFrame; // audio frame passed to webRTC AMP ProcessStream()
+ webrtc::AudioFrame* procFrame; // audio frame passed to webRTC AMP ProcessStream()
#else
// audio config strucutre
webrtc::AudioProcessing::Config config;
webrtc::StreamConfig inputConfig; // input stream configuration
webrtc::StreamConfig outputConfig; // output stream configuration
#endif
- int16_t *inBuf; // input buffer used when resampling
- size_t inBufSize; // input buffer size in frames
- size_t framesIn; // number of frames in input buffer
+ int16_t* inBuf; // input buffer used when resampling
+ size_t inBufSize; // input buffer size in frames
+ size_t framesIn; // number of frames in input buffer
#ifdef WEBRTC_LEGACY
- SpeexResamplerState *inResampler; // handle on input speex resampler
+ SpeexResamplerState* inResampler; // handle on input speex resampler
#endif
- int16_t *outBuf; // output buffer used when resampling
- size_t outBufSize; // output buffer size in frames
- size_t framesOut; // number of frames in output buffer
+ int16_t* outBuf; // output buffer used when resampling
+ size_t outBufSize; // output buffer size in frames
+ size_t framesOut; // number of frames in output buffer
#ifdef WEBRTC_LEGACY
- SpeexResamplerState *outResampler; // handle on output speex resampler
+ SpeexResamplerState* outResampler; // handle on output speex resampler
#endif
- uint32_t revChannelCount; // number of channels on reverse stream
- uint32_t revEnabledMsk; // bit field containing IDs of enabled pre processors
- // with reverse channel
- uint32_t revProcessedMsk; // bit field containing IDs of pre processors with reverse
- // channel already processed in current round
+ uint32_t revChannelCount; // number of channels on reverse stream
+ uint32_t revEnabledMsk; // bit field containing IDs of enabled pre processors
+ // with reverse channel
+ uint32_t revProcessedMsk; // bit field containing IDs of pre processors with reverse
+ // channel already processed in current round
#ifdef WEBRTC_LEGACY
- webrtc::AudioFrame *revFrame; // audio frame passed to webRTC AMP AnalyzeReverseStream()
+ webrtc::AudioFrame* revFrame; // audio frame passed to webRTC AMP AnalyzeReverseStream()
#else
webrtc::StreamConfig revConfig; // reverse stream configuration.
#endif
- int16_t *revBuf; // reverse channel input buffer
- size_t revBufSize; // reverse channel input buffer size
- size_t framesRev; // number of frames in reverse channel input buffer
+ int16_t* revBuf; // reverse channel input buffer
+ size_t revBufSize; // reverse channel input buffer size
+ size_t framesRev; // number of frames in reverse channel input buffer
#ifdef WEBRTC_LEGACY
- SpeexResamplerState *revResampler; // handle on reverse channel input speex resampler
+ SpeexResamplerState* revResampler; // handle on reverse channel input speex resampler
#endif
};
#ifdef DUAL_MIC_TEST
enum {
- PREPROC_CMD_DUAL_MIC_ENABLE = EFFECT_CMD_FIRST_PROPRIETARY, // enable dual mic mode
- PREPROC_CMD_DUAL_MIC_PCM_DUMP_START, // start pcm capture
- PREPROC_CMD_DUAL_MIC_PCM_DUMP_STOP // stop pcm capture
+ PREPROC_CMD_DUAL_MIC_ENABLE = EFFECT_CMD_FIRST_PROPRIETARY, // enable dual mic mode
+ PREPROC_CMD_DUAL_MIC_PCM_DUMP_START, // start pcm capture
+ PREPROC_CMD_DUAL_MIC_PCM_DUMP_STOP // stop pcm capture
};
enum {
@@ -180,24 +179,22 @@
};
const channel_config_t sDualMicConfigs[CHANNEL_CFG_CNT] = {
- {AUDIO_CHANNEL_IN_MONO , 0},
- {AUDIO_CHANNEL_IN_STEREO , 0},
- {AUDIO_CHANNEL_IN_FRONT , AUDIO_CHANNEL_IN_BACK},
- {AUDIO_CHANNEL_IN_STEREO , AUDIO_CHANNEL_IN_RIGHT}
-};
+ {AUDIO_CHANNEL_IN_MONO, 0},
+ {AUDIO_CHANNEL_IN_STEREO, 0},
+ {AUDIO_CHANNEL_IN_FRONT, AUDIO_CHANNEL_IN_BACK},
+ {AUDIO_CHANNEL_IN_STEREO, AUDIO_CHANNEL_IN_RIGHT}};
bool sHasAuxChannels[PREPROC_NUM_EFFECTS] = {
- false, // PREPROC_AGC
+ false, // PREPROC_AGC
true, // PREPROC_AEC
true, // PREPROC_NS
};
bool gDualMicEnabled;
-FILE *gPcmDumpFh;
+FILE* gPcmDumpFh;
static pthread_mutex_t gPcmDumpLock = PTHREAD_MUTEX_INITIALIZER;
#endif
-
//------------------------------------------------------------------------------
// Effect descriptors
//------------------------------------------------------------------------------
@@ -207,88 +204,75 @@
// Automatic Gain Control
static const effect_descriptor_t sAgcDescriptor = {
- { 0x0a8abfe0, 0x654c, 0x11e0, 0xba26, { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }, // type
- { 0xaa8130e0, 0x66fc, 0x11e0, 0xbad0, { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }, // uuid
+ {0x0a8abfe0, 0x654c, 0x11e0, 0xba26, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // type
+ {0xaa8130e0, 0x66fc, 0x11e0, 0xbad0, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // uuid
EFFECT_CONTROL_API_VERSION,
- (EFFECT_FLAG_TYPE_PRE_PROC|EFFECT_FLAG_DEVICE_IND),
- 0, //FIXME indicate CPU load
- 0, //FIXME indicate memory usage
+ (EFFECT_FLAG_TYPE_PRE_PROC | EFFECT_FLAG_DEVICE_IND),
+ 0, // FIXME indicate CPU load
+ 0, // FIXME indicate memory usage
"Automatic Gain Control",
- "The Android Open Source Project"
-};
+ "The Android Open Source Project"};
#ifndef WEBRTC_LEGACY
// Automatic Gain Control 2
static const effect_descriptor_t sAgc2Descriptor = {
- { 0xae3c653b, 0xbe18, 0x4ab8, 0x8938, { 0x41, 0x8f, 0x0a, 0x7f, 0x06, 0xac } }, // type
- { 0x89f38e65, 0xd4d2, 0x4d64, 0xad0e, { 0x2b, 0x3e, 0x79, 0x9e, 0xa8, 0x86 } }, // uuid
+ {0xae3c653b, 0xbe18, 0x4ab8, 0x8938, {0x41, 0x8f, 0x0a, 0x7f, 0x06, 0xac}}, // type
+ {0x89f38e65, 0xd4d2, 0x4d64, 0xad0e, {0x2b, 0x3e, 0x79, 0x9e, 0xa8, 0x86}}, // uuid
EFFECT_CONTROL_API_VERSION,
- (EFFECT_FLAG_TYPE_PRE_PROC|EFFECT_FLAG_DEVICE_IND),
- 0, //FIXME indicate CPU load
- 0, //FIXME indicate memory usage
+ (EFFECT_FLAG_TYPE_PRE_PROC | EFFECT_FLAG_DEVICE_IND),
+ 0, // FIXME indicate CPU load
+ 0, // FIXME indicate memory usage
"Automatic Gain Control 2",
- "The Android Open Source Project"
-};
+ "The Android Open Source Project"};
#endif
// Acoustic Echo Cancellation
static const effect_descriptor_t sAecDescriptor = {
- { 0x7b491460, 0x8d4d, 0x11e0, 0xbd61, { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }, // type
- { 0xbb392ec0, 0x8d4d, 0x11e0, 0xa896, { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }, // uuid
+ {0x7b491460, 0x8d4d, 0x11e0, 0xbd61, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // type
+ {0xbb392ec0, 0x8d4d, 0x11e0, 0xa896, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // uuid
EFFECT_CONTROL_API_VERSION,
- (EFFECT_FLAG_TYPE_PRE_PROC|EFFECT_FLAG_DEVICE_IND),
- 0, //FIXME indicate CPU load
- 0, //FIXME indicate memory usage
+ (EFFECT_FLAG_TYPE_PRE_PROC | EFFECT_FLAG_DEVICE_IND),
+ 0, // FIXME indicate CPU load
+ 0, // FIXME indicate memory usage
"Acoustic Echo Canceler",
- "The Android Open Source Project"
-};
+ "The Android Open Source Project"};
// Noise suppression
static const effect_descriptor_t sNsDescriptor = {
- { 0x58b4b260, 0x8e06, 0x11e0, 0xaa8e, { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }, // type
- { 0xc06c8400, 0x8e06, 0x11e0, 0x9cb6, { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }, // uuid
+ {0x58b4b260, 0x8e06, 0x11e0, 0xaa8e, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // type
+ {0xc06c8400, 0x8e06, 0x11e0, 0x9cb6, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // uuid
EFFECT_CONTROL_API_VERSION,
- (EFFECT_FLAG_TYPE_PRE_PROC|EFFECT_FLAG_DEVICE_IND),
- 0, //FIXME indicate CPU load
- 0, //FIXME indicate memory usage
+ (EFFECT_FLAG_TYPE_PRE_PROC | EFFECT_FLAG_DEVICE_IND),
+ 0, // FIXME indicate CPU load
+ 0, // FIXME indicate memory usage
"Noise Suppression",
- "The Android Open Source Project"
-};
+ "The Android Open Source Project"};
-
-static const effect_descriptor_t *sDescriptors[PREPROC_NUM_EFFECTS] = {
- &sAgcDescriptor,
+static const effect_descriptor_t* sDescriptors[PREPROC_NUM_EFFECTS] = {&sAgcDescriptor,
#ifndef WEBRTC_LEGACY
- &sAgc2Descriptor,
+ &sAgc2Descriptor,
#endif
- &sAecDescriptor,
- &sNsDescriptor
-};
+ &sAecDescriptor,
+ &sNsDescriptor};
//------------------------------------------------------------------------------
// Helper functions
//------------------------------------------------------------------------------
-const effect_uuid_t * const sUuidToPreProcTable[PREPROC_NUM_EFFECTS] = {
- FX_IID_AGC,
+const effect_uuid_t* const sUuidToPreProcTable[PREPROC_NUM_EFFECTS] = {FX_IID_AGC,
#ifndef WEBRTC_LEGACY
- FX_IID_AGC2,
+ FX_IID_AGC2,
#endif
- FX_IID_AEC,
- FX_IID_NS
-};
+ FX_IID_AEC, FX_IID_NS};
-
-const effect_uuid_t * ProcIdToUuid(int procId)
-{
+const effect_uuid_t* ProcIdToUuid(int procId) {
if (procId >= PREPROC_NUM_EFFECTS) {
return EFFECT_UUID_NULL;
}
return sUuidToPreProcTable[procId];
}
-uint32_t UuidToProcId(const effect_uuid_t * uuid)
-{
+uint32_t UuidToProcId(const effect_uuid_t* uuid) {
size_t i;
for (i = 0; i < PREPROC_NUM_EFFECTS; i++) {
if (memcmp(uuid, sUuidToPreProcTable[i], sizeof(*uuid)) == 0) {
@@ -298,15 +282,13 @@
return i;
}
-bool HasReverseStream(uint32_t procId)
-{
+bool HasReverseStream(uint32_t procId) {
if (procId == PREPROC_AEC) {
return true;
}
return false;
}
-
//------------------------------------------------------------------------------
// Automatic Gain Control (AGC)
//------------------------------------------------------------------------------
@@ -316,24 +298,22 @@
static const bool kAgcDefaultLimiter = true;
#ifndef WEBRTC_LEGACY
-int Agc2Init (preproc_effect_t *effect)
-{
+int Agc2Init(preproc_effect_t* effect) {
ALOGV("Agc2Init");
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.gain_controller2.fixed_digital.gain_db = 0.f;
effect->session->config.gain_controller2.adaptive_digital.level_estimator =
- effect->session->config.gain_controller2.kRms;
+ effect->session->config.gain_controller2.kRms;
effect->session->config.gain_controller2.adaptive_digital.extra_saturation_margin_db = 2.f;
effect->session->apm->ApplyConfig(effect->session->config);
return 0;
}
#endif
-int AgcInit (preproc_effect_t *effect)
-{
+int AgcInit(preproc_effect_t* effect) {
ALOGV("AgcInit");
#ifdef WEBRTC_LEGACY
- webrtc::GainControl *agc = static_cast<webrtc::GainControl *>(effect->engine);
+ webrtc::GainControl* agc = static_cast<webrtc::GainControl*>(effect->engine);
agc->set_mode(webrtc::GainControl::kFixedDigital);
agc->set_target_level_dbfs(kAgcDefaultTargetLevel);
agc->set_compression_gain_db(kAgcDefaultCompGain);
@@ -349,17 +329,15 @@
}
#ifndef WEBRTC_LEGACY
-int Agc2Create(preproc_effect_t *effect)
-{
+int Agc2Create(preproc_effect_t* effect) {
Agc2Init(effect);
return 0;
}
#endif
-int AgcCreate(preproc_effect_t *effect)
-{
+int AgcCreate(preproc_effect_t* effect) {
#ifdef WEBRTC_LEGACY
- webrtc::GainControl *agc = effect->session->apm->gain_control();
+ webrtc::GainControl* agc = effect->session->apm->gain_control();
ALOGV("AgcCreate got agc %p", agc);
if (agc == NULL) {
ALOGW("AgcCreate Error");
@@ -372,230 +350,216 @@
}
#ifndef WEBRTC_LEGACY
-int Agc2GetParameter(preproc_effect_t *effect,
- void *pParam,
- uint32_t *pValueSize,
- void *pValue)
-{
+int Agc2GetParameter(preproc_effect_t* effect, void* pParam, uint32_t* pValueSize, void* pValue) {
int status = 0;
- uint32_t param = *(uint32_t *)pParam;
- agc2_settings_t *pProperties = (agc2_settings_t *)pValue;
+ uint32_t param = *(uint32_t*)pParam;
+ agc2_settings_t* pProperties = (agc2_settings_t*)pValue;
switch (param) {
- case AGC2_PARAM_FIXED_DIGITAL_GAIN:
- if (*pValueSize < sizeof(float)) {
- *pValueSize = 0.f;
- return -EINVAL;
- }
- break;
- case AGC2_PARAM_ADAPT_DIGI_LEVEL_ESTIMATOR:
- if (*pValueSize < sizeof(int32_t)) {
- *pValueSize = 0;
- return -EINVAL;
- }
- break;
- case AGC2_PARAM_ADAPT_DIGI_EXTRA_SATURATION_MARGIN:
- if (*pValueSize < sizeof(float)) {
- *pValueSize = 0.f;
- return -EINVAL;
- }
- break;
- case AGC2_PARAM_PROPERTIES:
- if (*pValueSize < sizeof(agc2_settings_t)) {
- *pValueSize = 0;
- return -EINVAL;
- }
- break;
+ case AGC2_PARAM_FIXED_DIGITAL_GAIN:
+ if (*pValueSize < sizeof(float)) {
+ *pValueSize = 0.f;
+ return -EINVAL;
+ }
+ break;
+ case AGC2_PARAM_ADAPT_DIGI_LEVEL_ESTIMATOR:
+ if (*pValueSize < sizeof(int32_t)) {
+ *pValueSize = 0;
+ return -EINVAL;
+ }
+ break;
+ case AGC2_PARAM_ADAPT_DIGI_EXTRA_SATURATION_MARGIN:
+ if (*pValueSize < sizeof(float)) {
+ *pValueSize = 0.f;
+ return -EINVAL;
+ }
+ break;
+ case AGC2_PARAM_PROPERTIES:
+ if (*pValueSize < sizeof(agc2_settings_t)) {
+ *pValueSize = 0;
+ return -EINVAL;
+ }
+ break;
- default:
- ALOGW("Agc2GetParameter() unknown param %08x", param);
- status = -EINVAL;
- break;
+ default:
+ ALOGW("Agc2GetParameter() unknown param %08x", param);
+ status = -EINVAL;
+ break;
}
effect->session->config = effect->session->apm->GetConfig();
switch (param) {
- case AGC2_PARAM_FIXED_DIGITAL_GAIN:
- *(float *) pValue =
- (float)(effect->session->config.gain_controller2.fixed_digital.gain_db);
- ALOGV("Agc2GetParameter() target level %f dB", *(float *) pValue);
- break;
- case AGC2_PARAM_ADAPT_DIGI_LEVEL_ESTIMATOR:
- *(uint32_t *) pValue =
- (uint32_t)(effect->session->config.gain_controller2.adaptive_digital.
- level_estimator);
- ALOGV("Agc2GetParameter() level estimator %d",
- *(webrtc::AudioProcessing::Config::GainController2::LevelEstimator *) pValue);
- break;
- case AGC2_PARAM_ADAPT_DIGI_EXTRA_SATURATION_MARGIN:
- *(float *) pValue =
- (float)(effect->session->config.gain_controller2.adaptive_digital.
- extra_saturation_margin_db);
- ALOGV("Agc2GetParameter() extra saturation margin %f dB", *(float *) pValue);
- break;
- case AGC2_PARAM_PROPERTIES:
- pProperties->fixedDigitalGain =
- (float)(effect->session->config.gain_controller2.fixed_digital.gain_db);
- pProperties->level_estimator =
- (uint32_t)(effect->session->config.gain_controller2.adaptive_digital.
- level_estimator);
- pProperties->extraSaturationMargin =
- (float)(effect->session->config.gain_controller2.adaptive_digital.
- extra_saturation_margin_db);
- break;
- default:
- ALOGW("Agc2GetParameter() unknown param %d", param);
- status = -EINVAL;
- break;
+ case AGC2_PARAM_FIXED_DIGITAL_GAIN:
+ *(float*)pValue =
+ (float)(effect->session->config.gain_controller2.fixed_digital.gain_db);
+ ALOGV("Agc2GetParameter() target level %f dB", *(float*)pValue);
+ break;
+ case AGC2_PARAM_ADAPT_DIGI_LEVEL_ESTIMATOR:
+ *(uint32_t*)pValue = (uint32_t)(
+ effect->session->config.gain_controller2.adaptive_digital.level_estimator);
+ ALOGV("Agc2GetParameter() level estimator %d",
+ *(webrtc::AudioProcessing::Config::GainController2::LevelEstimator*)pValue);
+ break;
+ case AGC2_PARAM_ADAPT_DIGI_EXTRA_SATURATION_MARGIN:
+ *(float*)pValue = (float)(effect->session->config.gain_controller2.adaptive_digital
+ .extra_saturation_margin_db);
+ ALOGV("Agc2GetParameter() extra saturation margin %f dB", *(float*)pValue);
+ break;
+ case AGC2_PARAM_PROPERTIES:
+ pProperties->fixedDigitalGain =
+ (float)(effect->session->config.gain_controller2.fixed_digital.gain_db);
+ pProperties->level_estimator = (uint32_t)(
+ effect->session->config.gain_controller2.adaptive_digital.level_estimator);
+ pProperties->extraSaturationMargin =
+ (float)(effect->session->config.gain_controller2.adaptive_digital
+ .extra_saturation_margin_db);
+ break;
+ default:
+ ALOGW("Agc2GetParameter() unknown param %d", param);
+ status = -EINVAL;
+ break;
}
return status;
}
#endif
-int AgcGetParameter(preproc_effect_t *effect,
- void *pParam,
- uint32_t *pValueSize,
- void *pValue)
-{
+int AgcGetParameter(preproc_effect_t* effect, void* pParam, uint32_t* pValueSize, void* pValue) {
int status = 0;
- uint32_t param = *(uint32_t *)pParam;
- t_agc_settings *pProperties = (t_agc_settings *)pValue;
+ uint32_t param = *(uint32_t*)pParam;
+ t_agc_settings* pProperties = (t_agc_settings*)pValue;
#ifdef WEBRTC_LEGACY
- webrtc::GainControl *agc = static_cast<webrtc::GainControl *>(effect->engine);
+ webrtc::GainControl* agc = static_cast<webrtc::GainControl*>(effect->engine);
#endif
switch (param) {
- case AGC_PARAM_TARGET_LEVEL:
- case AGC_PARAM_COMP_GAIN:
- if (*pValueSize < sizeof(int16_t)) {
- *pValueSize = 0;
- return -EINVAL;
- }
- break;
- case AGC_PARAM_LIMITER_ENA:
- if (*pValueSize < sizeof(bool)) {
- *pValueSize = 0;
- return -EINVAL;
- }
- break;
- case AGC_PARAM_PROPERTIES:
- if (*pValueSize < sizeof(t_agc_settings)) {
- *pValueSize = 0;
- return -EINVAL;
- }
- break;
+ case AGC_PARAM_TARGET_LEVEL:
+ case AGC_PARAM_COMP_GAIN:
+ if (*pValueSize < sizeof(int16_t)) {
+ *pValueSize = 0;
+ return -EINVAL;
+ }
+ break;
+ case AGC_PARAM_LIMITER_ENA:
+ if (*pValueSize < sizeof(bool)) {
+ *pValueSize = 0;
+ return -EINVAL;
+ }
+ break;
+ case AGC_PARAM_PROPERTIES:
+ if (*pValueSize < sizeof(t_agc_settings)) {
+ *pValueSize = 0;
+ return -EINVAL;
+ }
+ break;
- default:
- ALOGW("AgcGetParameter() unknown param %08x", param);
- status = -EINVAL;
- break;
+ default:
+ ALOGW("AgcGetParameter() unknown param %08x", param);
+ status = -EINVAL;
+ break;
}
#ifdef WEBRTC_LEGACY
switch (param) {
- case AGC_PARAM_TARGET_LEVEL:
- *(int16_t *) pValue = (int16_t)(agc->target_level_dbfs() * -100);
- ALOGV("AgcGetParameter() target level %d milliBels", *(int16_t *) pValue);
- break;
- case AGC_PARAM_COMP_GAIN:
- *(int16_t *) pValue = (int16_t)(agc->compression_gain_db() * 100);
- ALOGV("AgcGetParameter() comp gain %d milliBels", *(int16_t *) pValue);
- break;
- case AGC_PARAM_LIMITER_ENA:
- *(bool *) pValue = (bool)agc->is_limiter_enabled();
- ALOGV("AgcGetParameter() limiter enabled %s",
- (*(int16_t *) pValue != 0) ? "true" : "false");
- break;
- case AGC_PARAM_PROPERTIES:
- pProperties->targetLevel = (int16_t)(agc->target_level_dbfs() * -100);
- pProperties->compGain = (int16_t)(agc->compression_gain_db() * 100);
- pProperties->limiterEnabled = (bool)agc->is_limiter_enabled();
- break;
- default:
- ALOGW("AgcGetParameter() unknown param %d", param);
- status = -EINVAL;
- break;
+ case AGC_PARAM_TARGET_LEVEL:
+ *(int16_t*)pValue = (int16_t)(agc->target_level_dbfs() * -100);
+ ALOGV("AgcGetParameter() target level %d milliBels", *(int16_t*)pValue);
+ break;
+ case AGC_PARAM_COMP_GAIN:
+ *(int16_t*)pValue = (int16_t)(agc->compression_gain_db() * 100);
+ ALOGV("AgcGetParameter() comp gain %d milliBels", *(int16_t*)pValue);
+ break;
+ case AGC_PARAM_LIMITER_ENA:
+ *(bool*)pValue = (bool)agc->is_limiter_enabled();
+ ALOGV("AgcGetParameter() limiter enabled %s",
+ (*(int16_t*)pValue != 0) ? "true" : "false");
+ break;
+ case AGC_PARAM_PROPERTIES:
+ pProperties->targetLevel = (int16_t)(agc->target_level_dbfs() * -100);
+ pProperties->compGain = (int16_t)(agc->compression_gain_db() * 100);
+ pProperties->limiterEnabled = (bool)agc->is_limiter_enabled();
+ break;
+ default:
+ ALOGW("AgcGetParameter() unknown param %d", param);
+ status = -EINVAL;
+ break;
}
#else
effect->session->config = effect->session->apm->GetConfig();
switch (param) {
- case AGC_PARAM_TARGET_LEVEL:
- *(int16_t *) pValue =
- (int16_t)(effect->session->config.gain_controller1.target_level_dbfs * -100);
- ALOGV("AgcGetParameter() target level %d milliBels", *(int16_t *) pValue);
- break;
- case AGC_PARAM_COMP_GAIN:
- *(int16_t *) pValue =
- (int16_t)(effect->session->config.gain_controller1.compression_gain_db * -100);
- ALOGV("AgcGetParameter() comp gain %d milliBels", *(int16_t *) pValue);
- break;
- case AGC_PARAM_LIMITER_ENA:
- *(bool *) pValue =
- (bool)(effect->session->config.gain_controller1.enable_limiter);
- ALOGV("AgcGetParameter() limiter enabled %s",
- (*(int16_t *) pValue != 0) ? "true" : "false");
- break;
- case AGC_PARAM_PROPERTIES:
- pProperties->targetLevel =
- (int16_t)(effect->session->config.gain_controller1.target_level_dbfs * -100);
- pProperties->compGain =
- (int16_t)(effect->session->config.gain_controller1.compression_gain_db * -100);
- pProperties->limiterEnabled =
- (bool)(effect->session->config.gain_controller1.enable_limiter);
- break;
- default:
- ALOGW("AgcGetParameter() unknown param %d", param);
- status = -EINVAL;
- break;
+ case AGC_PARAM_TARGET_LEVEL:
+ *(int16_t*)pValue =
+ (int16_t)(effect->session->config.gain_controller1.target_level_dbfs * -100);
+ ALOGV("AgcGetParameter() target level %d milliBels", *(int16_t*)pValue);
+ break;
+ case AGC_PARAM_COMP_GAIN:
+ *(int16_t*)pValue =
+ (int16_t)(effect->session->config.gain_controller1.compression_gain_db * -100);
+ ALOGV("AgcGetParameter() comp gain %d milliBels", *(int16_t*)pValue);
+ break;
+ case AGC_PARAM_LIMITER_ENA:
+ *(bool*)pValue = (bool)(effect->session->config.gain_controller1.enable_limiter);
+ ALOGV("AgcGetParameter() limiter enabled %s",
+ (*(int16_t*)pValue != 0) ? "true" : "false");
+ break;
+ case AGC_PARAM_PROPERTIES:
+ pProperties->targetLevel =
+ (int16_t)(effect->session->config.gain_controller1.target_level_dbfs * -100);
+ pProperties->compGain =
+ (int16_t)(effect->session->config.gain_controller1.compression_gain_db * -100);
+ pProperties->limiterEnabled =
+ (bool)(effect->session->config.gain_controller1.enable_limiter);
+ break;
+ default:
+ ALOGW("AgcGetParameter() unknown param %d", param);
+ status = -EINVAL;
+ break;
}
#endif
return status;
}
#ifndef WEBRTC_LEGACY
-int Agc2SetParameter (preproc_effect_t *effect, void *pParam, void *pValue)
-{
+int Agc2SetParameter(preproc_effect_t* effect, void* pParam, void* pValue) {
int status = 0;
- uint32_t param = *(uint32_t *)pParam;
+ uint32_t param = *(uint32_t*)pParam;
float valueFloat = 0.f;
- agc2_settings_t *pProperties = (agc2_settings_t *)pValue;
+ agc2_settings_t* pProperties = (agc2_settings_t*)pValue;
effect->session->config = effect->session->apm->GetConfig();
switch (param) {
- case AGC2_PARAM_FIXED_DIGITAL_GAIN:
- valueFloat = (float)(*(int32_t *) pValue);
- ALOGV("Agc2SetParameter() fixed digital gain %f dB", valueFloat);
- effect->session->config.gain_controller2.fixed_digital.gain_db = valueFloat;
- break;
- case AGC2_PARAM_ADAPT_DIGI_LEVEL_ESTIMATOR:
- ALOGV("Agc2SetParameter() level estimator %d", *(webrtc::AudioProcessing::Config::
- GainController2::LevelEstimator *) pValue);
- effect->session->config.gain_controller2.adaptive_digital.level_estimator =
- (*(webrtc::AudioProcessing::Config::GainController2::LevelEstimator *) pValue);
- break;
- case AGC2_PARAM_ADAPT_DIGI_EXTRA_SATURATION_MARGIN:
- valueFloat = (float)(*(int32_t *) pValue);
- ALOGV("Agc2SetParameter() extra saturation margin %f dB", valueFloat);
- effect->session->config.gain_controller2.adaptive_digital.extra_saturation_margin_db =
- valueFloat;
- break;
- case AGC2_PARAM_PROPERTIES:
- ALOGV("Agc2SetParameter() properties gain %f, level %d margin %f",
- pProperties->fixedDigitalGain,
- pProperties->level_estimator,
- pProperties->extraSaturationMargin);
- effect->session->config.gain_controller2.fixed_digital.gain_db =
- pProperties->fixedDigitalGain;
- effect->session->config.gain_controller2.adaptive_digital.level_estimator =
- (webrtc::AudioProcessing::Config::GainController2::LevelEstimator)pProperties->
- level_estimator;
- effect->session->config.gain_controller2.adaptive_digital.extra_saturation_margin_db =
- pProperties->extraSaturationMargin;
- break;
- default:
- ALOGW("Agc2SetParameter() unknown param %08x value %08x", param, *(uint32_t *)pValue);
- status = -EINVAL;
- break;
+ case AGC2_PARAM_FIXED_DIGITAL_GAIN:
+ valueFloat = (float)(*(int32_t*)pValue);
+ ALOGV("Agc2SetParameter() fixed digital gain %f dB", valueFloat);
+ effect->session->config.gain_controller2.fixed_digital.gain_db = valueFloat;
+ break;
+ case AGC2_PARAM_ADAPT_DIGI_LEVEL_ESTIMATOR:
+ ALOGV("Agc2SetParameter() level estimator %d",
+ *(webrtc::AudioProcessing::Config::GainController2::LevelEstimator*)pValue);
+ effect->session->config.gain_controller2.adaptive_digital.level_estimator =
+ (*(webrtc::AudioProcessing::Config::GainController2::LevelEstimator*)pValue);
+ break;
+ case AGC2_PARAM_ADAPT_DIGI_EXTRA_SATURATION_MARGIN:
+ valueFloat = (float)(*(int32_t*)pValue);
+ ALOGV("Agc2SetParameter() extra saturation margin %f dB", valueFloat);
+ effect->session->config.gain_controller2.adaptive_digital.extra_saturation_margin_db =
+ valueFloat;
+ break;
+ case AGC2_PARAM_PROPERTIES:
+ ALOGV("Agc2SetParameter() properties gain %f, level %d margin %f",
+ pProperties->fixedDigitalGain, pProperties->level_estimator,
+ pProperties->extraSaturationMargin);
+ effect->session->config.gain_controller2.fixed_digital.gain_db =
+ pProperties->fixedDigitalGain;
+ effect->session->config.gain_controller2.adaptive_digital.level_estimator =
+ (webrtc::AudioProcessing::Config::GainController2::LevelEstimator)
+ pProperties->level_estimator;
+ effect->session->config.gain_controller2.adaptive_digital.extra_saturation_margin_db =
+ pProperties->extraSaturationMargin;
+ break;
+ default:
+ ALOGW("Agc2SetParameter() unknown param %08x value %08x", param, *(uint32_t*)pValue);
+ status = -EINVAL;
+ break;
}
effect->session->apm->ApplyConfig(effect->session->config);
@@ -605,79 +569,72 @@
}
#endif
-int AgcSetParameter (preproc_effect_t *effect, void *pParam, void *pValue)
-{
+int AgcSetParameter(preproc_effect_t* effect, void* pParam, void* pValue) {
int status = 0;
#ifdef WEBRTC_LEGACY
- uint32_t param = *(uint32_t *)pParam;
- t_agc_settings *pProperties = (t_agc_settings *)pValue;
- webrtc::GainControl *agc = static_cast<webrtc::GainControl *>(effect->engine);
+ uint32_t param = *(uint32_t*)pParam;
+ t_agc_settings* pProperties = (t_agc_settings*)pValue;
+ webrtc::GainControl* agc = static_cast<webrtc::GainControl*>(effect->engine);
switch (param) {
- case AGC_PARAM_TARGET_LEVEL:
- ALOGV("AgcSetParameter() target level %d milliBels", *(int16_t *)pValue);
- status = agc->set_target_level_dbfs(-(*(int16_t *)pValue / 100));
- break;
- case AGC_PARAM_COMP_GAIN:
- ALOGV("AgcSetParameter() comp gain %d milliBels", *(int16_t *)pValue);
- status = agc->set_compression_gain_db(*(int16_t *)pValue / 100);
- break;
- case AGC_PARAM_LIMITER_ENA:
- ALOGV("AgcSetParameter() limiter enabled %s", *(bool *)pValue ? "true" : "false");
- status = agc->enable_limiter(*(bool *)pValue);
- break;
- case AGC_PARAM_PROPERTIES:
- ALOGV("AgcSetParameter() properties level %d, gain %d limiter %d",
- pProperties->targetLevel,
- pProperties->compGain,
- pProperties->limiterEnabled);
- status = agc->set_target_level_dbfs(-(pProperties->targetLevel / 100));
- if (status != 0) break;
- status = agc->set_compression_gain_db(pProperties->compGain / 100);
- if (status != 0) break;
- status = agc->enable_limiter(pProperties->limiterEnabled);
- break;
- default:
- ALOGW("AgcSetParameter() unknown param %08x value %08x", param, *(uint32_t *)pValue);
- status = -EINVAL;
- break;
+ case AGC_PARAM_TARGET_LEVEL:
+ ALOGV("AgcSetParameter() target level %d milliBels", *(int16_t*)pValue);
+ status = agc->set_target_level_dbfs(-(*(int16_t*)pValue / 100));
+ break;
+ case AGC_PARAM_COMP_GAIN:
+ ALOGV("AgcSetParameter() comp gain %d milliBels", *(int16_t*)pValue);
+ status = agc->set_compression_gain_db(*(int16_t*)pValue / 100);
+ break;
+ case AGC_PARAM_LIMITER_ENA:
+ ALOGV("AgcSetParameter() limiter enabled %s", *(bool*)pValue ? "true" : "false");
+ status = agc->enable_limiter(*(bool*)pValue);
+ break;
+ case AGC_PARAM_PROPERTIES:
+ ALOGV("AgcSetParameter() properties level %d, gain %d limiter %d",
+ pProperties->targetLevel, pProperties->compGain, pProperties->limiterEnabled);
+ status = agc->set_target_level_dbfs(-(pProperties->targetLevel / 100));
+ if (status != 0) break;
+ status = agc->set_compression_gain_db(pProperties->compGain / 100);
+ if (status != 0) break;
+ status = agc->enable_limiter(pProperties->limiterEnabled);
+ break;
+ default:
+ ALOGW("AgcSetParameter() unknown param %08x value %08x", param, *(uint32_t*)pValue);
+ status = -EINVAL;
+ break;
}
#else
- uint32_t param = *(uint32_t *)pParam;
- t_agc_settings *pProperties = (t_agc_settings *)pValue;
+ uint32_t param = *(uint32_t*)pParam;
+ t_agc_settings* pProperties = (t_agc_settings*)pValue;
effect->session->config = effect->session->apm->GetConfig();
switch (param) {
- case AGC_PARAM_TARGET_LEVEL:
- ALOGV("AgcSetParameter() target level %d milliBels", *(int16_t *)pValue);
- effect->session->config.gain_controller1.target_level_dbfs =
- (-(*(int16_t *)pValue / 100));
- break;
- case AGC_PARAM_COMP_GAIN:
- ALOGV("AgcSetParameter() comp gain %d milliBels", *(int16_t *)pValue);
- effect->session->config.gain_controller1.compression_gain_db =
- (*(int16_t *)pValue / 100);
- break;
- case AGC_PARAM_LIMITER_ENA:
- ALOGV("AgcSetParameter() limiter enabled %s", *(bool *)pValue ? "true" : "false");
- effect->session->config.gain_controller1.enable_limiter =
- (*(bool *)pValue);
- break;
- case AGC_PARAM_PROPERTIES:
- ALOGV("AgcSetParameter() properties level %d, gain %d limiter %d",
- pProperties->targetLevel,
- pProperties->compGain,
- pProperties->limiterEnabled);
- effect->session->config.gain_controller1.target_level_dbfs =
- -(pProperties->targetLevel / 100);
- effect->session->config.gain_controller1.compression_gain_db =
- pProperties->compGain / 100;
- effect->session->config.gain_controller1.enable_limiter =
- pProperties->limiterEnabled;
- break;
- default:
- ALOGW("AgcSetParameter() unknown param %08x value %08x", param, *(uint32_t *)pValue);
- status = -EINVAL;
- break;
+ case AGC_PARAM_TARGET_LEVEL:
+ ALOGV("AgcSetParameter() target level %d milliBels", *(int16_t*)pValue);
+ effect->session->config.gain_controller1.target_level_dbfs =
+ (-(*(int16_t*)pValue / 100));
+ break;
+ case AGC_PARAM_COMP_GAIN:
+ ALOGV("AgcSetParameter() comp gain %d milliBels", *(int16_t*)pValue);
+ effect->session->config.gain_controller1.compression_gain_db =
+ (*(int16_t*)pValue / 100);
+ break;
+ case AGC_PARAM_LIMITER_ENA:
+ ALOGV("AgcSetParameter() limiter enabled %s", *(bool*)pValue ? "true" : "false");
+ effect->session->config.gain_controller1.enable_limiter = (*(bool*)pValue);
+ break;
+ case AGC_PARAM_PROPERTIES:
+ ALOGV("AgcSetParameter() properties level %d, gain %d limiter %d",
+ pProperties->targetLevel, pProperties->compGain, pProperties->limiterEnabled);
+ effect->session->config.gain_controller1.target_level_dbfs =
+ -(pProperties->targetLevel / 100);
+ effect->session->config.gain_controller1.compression_gain_db =
+ pProperties->compGain / 100;
+ effect->session->config.gain_controller1.enable_limiter = pProperties->limiterEnabled;
+ break;
+ default:
+ ALOGW("AgcSetParameter() unknown param %08x value %08x", param, *(uint32_t*)pValue);
+ status = -EINVAL;
+ break;
}
effect->session->apm->ApplyConfig(effect->session->config);
#endif
@@ -688,18 +645,16 @@
}
#ifndef WEBRTC_LEGACY
-void Agc2Enable(preproc_effect_t *effect)
-{
+void Agc2Enable(preproc_effect_t* effect) {
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.gain_controller2.enabled = true;
effect->session->apm->ApplyConfig(effect->session->config);
}
#endif
-void AgcEnable(preproc_effect_t *effect)
-{
+void AgcEnable(preproc_effect_t* effect) {
#ifdef WEBRTC_LEGACY
- webrtc::GainControl *agc = static_cast<webrtc::GainControl *>(effect->engine);
+ webrtc::GainControl* agc = static_cast<webrtc::GainControl*>(effect->engine);
ALOGV("AgcEnable agc %p", agc);
agc->Enable(true);
#else
@@ -710,19 +665,17 @@
}
#ifndef WEBRTC_LEGACY
-void Agc2Disable(preproc_effect_t *effect)
-{
+void Agc2Disable(preproc_effect_t* effect) {
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.gain_controller2.enabled = false;
effect->session->apm->ApplyConfig(effect->session->config);
}
#endif
-void AgcDisable(preproc_effect_t *effect)
-{
+void AgcDisable(preproc_effect_t* effect) {
#ifdef WEBRTC_LEGACY
ALOGV("AgcDisable");
- webrtc::GainControl *agc = static_cast<webrtc::GainControl *>(effect->engine);
+ webrtc::GainControl* agc = static_cast<webrtc::GainControl*>(effect->engine);
agc->Enable(false);
#else
effect->session->config = effect->session->apm->GetConfig();
@@ -731,28 +684,13 @@
#endif
}
-static const preproc_ops_t sAgcOps = {
- AgcCreate,
- AgcInit,
- NULL,
- AgcEnable,
- AgcDisable,
- AgcSetParameter,
- AgcGetParameter,
- NULL
-};
+static const preproc_ops_t sAgcOps = {AgcCreate, AgcInit, NULL, AgcEnable, AgcDisable,
+ AgcSetParameter, AgcGetParameter, NULL};
#ifndef WEBRTC_LEGACY
-static const preproc_ops_t sAgc2Ops = {
- Agc2Create,
- Agc2Init,
- NULL,
- Agc2Enable,
- Agc2Disable,
- Agc2SetParameter,
- Agc2GetParameter,
- NULL
-};
+static const preproc_ops_t sAgc2Ops = {Agc2Create, Agc2Init, NULL,
+ Agc2Enable, Agc2Disable, Agc2SetParameter,
+ Agc2GetParameter, NULL};
#endif
//------------------------------------------------------------------------------
@@ -765,26 +703,23 @@
static const bool kAecDefaultComfortNoise = true;
#endif
-int AecInit (preproc_effect_t *effect)
-{
+int AecInit(preproc_effect_t* effect) {
ALOGV("AecInit");
#ifdef WEBRTC_LEGACY
- webrtc::EchoControlMobile *aec = static_cast<webrtc::EchoControlMobile *>(effect->engine);
+ webrtc::EchoControlMobile* aec = static_cast<webrtc::EchoControlMobile*>(effect->engine);
aec->set_routing_mode(kAecDefaultMode);
aec->enable_comfort_noise(kAecDefaultComfortNoise);
#else
- effect->session->config =
- effect->session->apm->GetConfig() ;
- effect->session->config.echo_canceller.mobile_mode = false;
+ effect->session->config = effect->session->apm->GetConfig();
+ effect->session->config.echo_canceller.mobile_mode = true;
effect->session->apm->ApplyConfig(effect->session->config);
#endif
return 0;
}
-int AecCreate(preproc_effect_t *effect)
-{
+int AecCreate(preproc_effect_t* effect) {
#ifdef WEBRTC_LEGACY
- webrtc::EchoControlMobile *aec = effect->session->apm->echo_control_mobile();
+ webrtc::EchoControlMobile* aec = effect->session->apm->echo_control_mobile();
ALOGV("AecCreate got aec %p", aec);
if (aec == NULL) {
ALOGW("AgcCreate Error");
@@ -792,76 +727,68 @@
}
effect->engine = static_cast<preproc_fx_handle_t>(aec);
#endif
- AecInit (effect);
+ AecInit(effect);
return 0;
}
-int AecGetParameter(preproc_effect_t *effect,
- void *pParam,
- uint32_t *pValueSize,
- void *pValue)
-{
+int AecGetParameter(preproc_effect_t* effect, void* pParam, uint32_t* pValueSize, void* pValue) {
int status = 0;
- uint32_t param = *(uint32_t *)pParam;
+ uint32_t param = *(uint32_t*)pParam;
if (*pValueSize < sizeof(uint32_t)) {
return -EINVAL;
}
switch (param) {
- case AEC_PARAM_ECHO_DELAY:
- case AEC_PARAM_PROPERTIES:
- *(uint32_t *)pValue = 1000 * effect->session->apm->stream_delay_ms();
- ALOGV("AecGetParameter() echo delay %d us", *(uint32_t *)pValue);
- break;
+ case AEC_PARAM_ECHO_DELAY:
+ case AEC_PARAM_PROPERTIES:
+ *(uint32_t*)pValue = 1000 * effect->session->apm->stream_delay_ms();
+ ALOGV("AecGetParameter() echo delay %d us", *(uint32_t*)pValue);
+ break;
#ifndef WEBRTC_LEGACY
- case AEC_PARAM_MOBILE_MODE:
- effect->session->config =
- effect->session->apm->GetConfig() ;
- *(uint32_t *)pValue = effect->session->config.echo_canceller.mobile_mode;
- ALOGV("AecGetParameter() mobile mode %d us", *(uint32_t *)pValue);
- break;
+ case AEC_PARAM_MOBILE_MODE:
+ effect->session->config = effect->session->apm->GetConfig();
+ *(uint32_t*)pValue = effect->session->config.echo_canceller.mobile_mode;
+ ALOGV("AecGetParameter() mobile mode %d us", *(uint32_t*)pValue);
+ break;
#endif
- default:
- ALOGW("AecGetParameter() unknown param %08x value %08x", param, *(uint32_t *)pValue);
- status = -EINVAL;
- break;
+ default:
+ ALOGW("AecGetParameter() unknown param %08x value %08x", param, *(uint32_t*)pValue);
+ status = -EINVAL;
+ break;
}
return status;
}
-int AecSetParameter (preproc_effect_t *effect, void *pParam, void *pValue)
-{
+int AecSetParameter(preproc_effect_t* effect, void* pParam, void* pValue) {
int status = 0;
- uint32_t param = *(uint32_t *)pParam;
- uint32_t value = *(uint32_t *)pValue;
+ uint32_t param = *(uint32_t*)pParam;
+ uint32_t value = *(uint32_t*)pValue;
switch (param) {
- case AEC_PARAM_ECHO_DELAY:
- case AEC_PARAM_PROPERTIES:
- status = effect->session->apm->set_stream_delay_ms(value/1000);
- ALOGV("AecSetParameter() echo delay %d us, status %d", value, status);
- break;
+ case AEC_PARAM_ECHO_DELAY:
+ case AEC_PARAM_PROPERTIES:
+ status = effect->session->apm->set_stream_delay_ms(value / 1000);
+ ALOGV("AecSetParameter() echo delay %d us, status %d", value, status);
+ break;
#ifndef WEBRTC_LEGACY
- case AEC_PARAM_MOBILE_MODE:
- effect->session->config =
- effect->session->apm->GetConfig() ;
- effect->session->config.echo_canceller.mobile_mode = value;
- ALOGV("AecSetParameter() mobile mode %d us", value);
- effect->session->apm->ApplyConfig(effect->session->config);
- break;
+ case AEC_PARAM_MOBILE_MODE:
+ effect->session->config = effect->session->apm->GetConfig();
+ effect->session->config.echo_canceller.mobile_mode = value;
+ ALOGV("AecSetParameter() mobile mode %d us", value);
+ effect->session->apm->ApplyConfig(effect->session->config);
+ break;
#endif
- default:
- ALOGW("AecSetParameter() unknown param %08x value %08x", param, *(uint32_t *)pValue);
- status = -EINVAL;
- break;
+ default:
+ ALOGW("AecSetParameter() unknown param %08x value %08x", param, *(uint32_t*)pValue);
+ status = -EINVAL;
+ break;
}
return status;
}
-void AecEnable(preproc_effect_t *effect)
-{
+void AecEnable(preproc_effect_t* effect) {
#ifdef WEBRTC_LEGACY
- webrtc::EchoControlMobile *aec = static_cast<webrtc::EchoControlMobile *>(effect->engine);
+ webrtc::EchoControlMobile* aec = static_cast<webrtc::EchoControlMobile*>(effect->engine);
ALOGV("AecEnable aec %p", aec);
aec->Enable(true);
#else
@@ -871,11 +798,10 @@
#endif
}
-void AecDisable(preproc_effect_t *effect)
-{
+void AecDisable(preproc_effect_t* effect) {
#ifdef WEBRTC_LEGACY
ALOGV("AecDisable");
- webrtc::EchoControlMobile *aec = static_cast<webrtc::EchoControlMobile *>(effect->engine);
+ webrtc::EchoControlMobile* aec = static_cast<webrtc::EchoControlMobile*>(effect->engine);
aec->Enable(false);
#else
effect->session->config = effect->session->apm->GetConfig();
@@ -884,12 +810,12 @@
#endif
}
-int AecSetDevice(preproc_effect_t *effect, uint32_t device)
-{
+int AecSetDevice(preproc_effect_t* effect, uint32_t device) {
ALOGV("AecSetDevice %08x", device);
#ifdef WEBRTC_LEGACY
- webrtc::EchoControlMobile *aec = static_cast<webrtc::EchoControlMobile *>(effect->engine);
- webrtc::EchoControlMobile::RoutingMode mode = webrtc::EchoControlMobile::kQuietEarpieceOrHeadset;
+ webrtc::EchoControlMobile* aec = static_cast<webrtc::EchoControlMobile*>(effect->engine);
+ webrtc::EchoControlMobile::RoutingMode mode =
+ webrtc::EchoControlMobile::kQuietEarpieceOrHeadset;
#endif
if (audio_is_input_device(device)) {
@@ -897,34 +823,27 @@
}
#ifdef WEBRTC_LEGACY
- switch(device) {
- case AUDIO_DEVICE_OUT_EARPIECE:
- mode = webrtc::EchoControlMobile::kEarpiece;
- break;
- case AUDIO_DEVICE_OUT_SPEAKER:
- mode = webrtc::EchoControlMobile::kSpeakerphone;
- break;
- case AUDIO_DEVICE_OUT_WIRED_HEADSET:
- case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
- case AUDIO_DEVICE_OUT_USB_HEADSET:
- default:
- break;
+ switch (device) {
+ case AUDIO_DEVICE_OUT_EARPIECE:
+ mode = webrtc::EchoControlMobile::kEarpiece;
+ break;
+ case AUDIO_DEVICE_OUT_SPEAKER:
+ mode = webrtc::EchoControlMobile::kSpeakerphone;
+ break;
+ case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+ case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+ case AUDIO_DEVICE_OUT_USB_HEADSET:
+ default:
+ break;
}
aec->set_routing_mode(mode);
#endif
return 0;
}
-static const preproc_ops_t sAecOps = {
- AecCreate,
- AecInit,
- NULL,
- AecEnable,
- AecDisable,
- AecSetParameter,
- AecGetParameter,
- AecSetDevice
-};
+static const preproc_ops_t sAecOps = {AecCreate, AecInit, NULL,
+ AecEnable, AecDisable, AecSetParameter,
+ AecGetParameter, AecSetDevice};
//------------------------------------------------------------------------------
// Noise Suppression (NS)
@@ -934,14 +853,13 @@
static const webrtc::NoiseSuppression::Level kNsDefaultLevel = webrtc::NoiseSuppression::kModerate;
#else
static const webrtc::AudioProcessing::Config::NoiseSuppression::Level kNsDefaultLevel =
- webrtc::AudioProcessing::Config::NoiseSuppression::kModerate;
+ webrtc::AudioProcessing::Config::NoiseSuppression::kModerate;
#endif
-int NsInit (preproc_effect_t *effect)
-{
+int NsInit(preproc_effect_t* effect) {
ALOGV("NsInit");
#ifdef WEBRTC_LEGACY
- webrtc::NoiseSuppression *ns = static_cast<webrtc::NoiseSuppression *>(effect->engine);
+ webrtc::NoiseSuppression* ns = static_cast<webrtc::NoiseSuppression*>(effect->engine);
ns->set_level(kNsDefaultLevel);
webrtc::Config config;
std::vector<webrtc::Point> geometry;
@@ -951,27 +869,22 @@
geometry.push_back(webrtc::Point(0.01f, 0.f, 0.f));
geometry.push_back(webrtc::Point(0.03f, 0.f, 0.f));
// The geometry needs to be set with Beamforming enabled.
- config.Set<webrtc::Beamforming>(
- new webrtc::Beamforming(true, geometry));
+ config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry));
effect->session->apm->SetExtraOptions(config);
- config.Set<webrtc::Beamforming>(
- new webrtc::Beamforming(false, geometry));
+ config.Set<webrtc::Beamforming>(new webrtc::Beamforming(false, geometry));
effect->session->apm->SetExtraOptions(config);
#else
- effect->session->config =
- effect->session->apm->GetConfig() ;
- effect->session->config.noise_suppression.level =
- kNsDefaultLevel;
+ effect->session->config = effect->session->apm->GetConfig();
+ effect->session->config.noise_suppression.level = kNsDefaultLevel;
effect->session->apm->ApplyConfig(effect->session->config);
#endif
effect->type = NS_TYPE_SINGLE_CHANNEL;
return 0;
}
-int NsCreate(preproc_effect_t *effect)
-{
+int NsCreate(preproc_effect_t* effect) {
#ifdef WEBRTC_LEGACY
- webrtc::NoiseSuppression *ns = effect->session->apm->noise_suppression();
+ webrtc::NoiseSuppression* ns = effect->session->apm->noise_suppression();
ALOGV("NsCreate got ns %p", ns);
if (ns == NULL) {
ALOGW("AgcCreate Error");
@@ -979,37 +892,31 @@
}
effect->engine = static_cast<preproc_fx_handle_t>(ns);
#endif
- NsInit (effect);
+ NsInit(effect);
return 0;
}
-int NsGetParameter(preproc_effect_t *effect __unused,
- void *pParam __unused,
- uint32_t *pValueSize __unused,
- void *pValue __unused)
-{
+int NsGetParameter(preproc_effect_t* effect __unused, void* pParam __unused,
+ uint32_t* pValueSize __unused, void* pValue __unused) {
int status = 0;
return status;
}
-int NsSetParameter (preproc_effect_t *effect, void *pParam, void *pValue)
-{
+int NsSetParameter(preproc_effect_t* effect, void* pParam, void* pValue) {
int status = 0;
#ifdef WEBRTC_LEGACY
- webrtc::NoiseSuppression *ns = static_cast<webrtc::NoiseSuppression *>(effect->engine);
- uint32_t param = *(uint32_t *)pParam;
- uint32_t value = *(uint32_t *)pValue;
- switch(param) {
+ webrtc::NoiseSuppression* ns = static_cast<webrtc::NoiseSuppression*>(effect->engine);
+ uint32_t param = *(uint32_t*)pParam;
+ uint32_t value = *(uint32_t*)pValue;
+ switch (param) {
case NS_PARAM_LEVEL:
ns->set_level((webrtc::NoiseSuppression::Level)value);
ALOGV("NsSetParameter() level %d", value);
break;
- case NS_PARAM_TYPE:
- {
+ case NS_PARAM_TYPE: {
webrtc::Config config;
std::vector<webrtc::Point> geometry;
- bool is_beamforming_enabled =
- value == NS_TYPE_MULTI_CHANNEL && ns->is_enabled();
+ bool is_beamforming_enabled = value == NS_TYPE_MULTI_CHANNEL && ns->is_enabled();
config.Set<webrtc::Beamforming>(
new webrtc::Beamforming(is_beamforming_enabled, geometry));
effect->session->apm->SetExtraOptions(config);
@@ -1022,14 +929,13 @@
status = -EINVAL;
}
#else
- uint32_t param = *(uint32_t *)pParam;
- uint32_t value = *(uint32_t *)pValue;
- effect->session->config =
- effect->session->apm->GetConfig();
+ uint32_t param = *(uint32_t*)pParam;
+ uint32_t value = *(uint32_t*)pValue;
+ effect->session->config = effect->session->apm->GetConfig();
switch (param) {
case NS_PARAM_LEVEL:
effect->session->config.noise_suppression.level =
- (webrtc::AudioProcessing::Config::NoiseSuppression::Level)value;
+ (webrtc::AudioProcessing::Config::NoiseSuppression::Level)value;
ALOGV("NsSetParameter() level %d", value);
break;
default:
@@ -1042,10 +948,9 @@
return status;
}
-void NsEnable(preproc_effect_t *effect)
-{
+void NsEnable(preproc_effect_t* effect) {
#ifdef WEBRTC_LEGACY
- webrtc::NoiseSuppression *ns = static_cast<webrtc::NoiseSuppression *>(effect->engine);
+ webrtc::NoiseSuppression* ns = static_cast<webrtc::NoiseSuppression*>(effect->engine);
ALOGV("NsEnable ns %p", ns);
ns->Enable(true);
if (effect->type == NS_TYPE_MULTI_CHANNEL) {
@@ -1055,137 +960,118 @@
effect->session->apm->SetExtraOptions(config);
}
#else
- effect->session->config =
- effect->session->apm->GetConfig();
+ effect->session->config = effect->session->apm->GetConfig();
effect->session->config.noise_suppression.enabled = true;
effect->session->apm->ApplyConfig(effect->session->config);
#endif
}
-void NsDisable(preproc_effect_t *effect)
-{
+void NsDisable(preproc_effect_t* effect) {
ALOGV("NsDisable");
#ifdef WEBRTC_LEGACY
- webrtc::NoiseSuppression *ns = static_cast<webrtc::NoiseSuppression *>(effect->engine);
+ webrtc::NoiseSuppression* ns = static_cast<webrtc::NoiseSuppression*>(effect->engine);
ns->Enable(false);
webrtc::Config config;
std::vector<webrtc::Point> geometry;
config.Set<webrtc::Beamforming>(new webrtc::Beamforming(false, geometry));
effect->session->apm->SetExtraOptions(config);
#else
- effect->session->config =
- effect->session->apm->GetConfig();
+ effect->session->config = effect->session->apm->GetConfig();
effect->session->config.noise_suppression.enabled = false;
effect->session->apm->ApplyConfig(effect->session->config);
#endif
}
-static const preproc_ops_t sNsOps = {
- NsCreate,
- NsInit,
- NULL,
- NsEnable,
- NsDisable,
- NsSetParameter,
- NsGetParameter,
- NULL
-};
+static const preproc_ops_t sNsOps = {NsCreate, NsInit, NULL, NsEnable,
+ NsDisable, NsSetParameter, NsGetParameter, NULL};
-
-
-static const preproc_ops_t *sPreProcOps[PREPROC_NUM_EFFECTS] = {
- &sAgcOps,
+static const preproc_ops_t* sPreProcOps[PREPROC_NUM_EFFECTS] = {&sAgcOps,
#ifndef WEBRTC_LEGACY
- &sAgc2Ops,
+ &sAgc2Ops,
#endif
- &sAecOps,
- &sNsOps
-};
-
+ &sAecOps, &sNsOps};
//------------------------------------------------------------------------------
// Effect functions
//------------------------------------------------------------------------------
-void Session_SetProcEnabled(preproc_session_t *session, uint32_t procId, bool enabled);
+void Session_SetProcEnabled(preproc_session_t* session, uint32_t procId, bool enabled);
extern "C" const struct effect_interface_s sEffectInterface;
extern "C" const struct effect_interface_s sEffectInterfaceReverse;
-#define BAD_STATE_ABORT(from, to) \
- LOG_ALWAYS_FATAL("Bad state transition from %d to %d", from, to);
+#define BAD_STATE_ABORT(from, to) LOG_ALWAYS_FATAL("Bad state transition from %d to %d", from, to);
-int Effect_SetState(preproc_effect_t *effect, uint32_t state)
-{
+int Effect_SetState(preproc_effect_t* effect, uint32_t state) {
int status = 0;
ALOGV("Effect_SetState proc %d, new %d old %d", effect->procId, state, effect->state);
- switch(state) {
- case PREPROC_EFFECT_STATE_INIT:
- switch(effect->state) {
- case PREPROC_EFFECT_STATE_ACTIVE:
- effect->ops->disable(effect);
- Session_SetProcEnabled(effect->session, effect->procId, false);
+ switch (state) {
+ case PREPROC_EFFECT_STATE_INIT:
+ switch (effect->state) {
+ case PREPROC_EFFECT_STATE_ACTIVE:
+ effect->ops->disable(effect);
+ Session_SetProcEnabled(effect->session, effect->procId, false);
+ break;
+ case PREPROC_EFFECT_STATE_CONFIG:
+ case PREPROC_EFFECT_STATE_CREATED:
+ case PREPROC_EFFECT_STATE_INIT:
+ break;
+ default:
+ BAD_STATE_ABORT(effect->state, state);
+ }
+ break;
+ case PREPROC_EFFECT_STATE_CREATED:
+ switch (effect->state) {
+ case PREPROC_EFFECT_STATE_INIT:
+ status = effect->ops->create(effect);
+ break;
+ case PREPROC_EFFECT_STATE_CREATED:
+ case PREPROC_EFFECT_STATE_ACTIVE:
+ case PREPROC_EFFECT_STATE_CONFIG:
+ ALOGE("Effect_SetState invalid transition");
+ status = -ENOSYS;
+ break;
+ default:
+ BAD_STATE_ABORT(effect->state, state);
+ }
break;
case PREPROC_EFFECT_STATE_CONFIG:
- case PREPROC_EFFECT_STATE_CREATED:
- case PREPROC_EFFECT_STATE_INIT:
+ switch (effect->state) {
+ case PREPROC_EFFECT_STATE_INIT:
+ ALOGE("Effect_SetState invalid transition");
+ status = -ENOSYS;
+ break;
+ case PREPROC_EFFECT_STATE_ACTIVE:
+ effect->ops->disable(effect);
+ Session_SetProcEnabled(effect->session, effect->procId, false);
+ break;
+ case PREPROC_EFFECT_STATE_CREATED:
+ case PREPROC_EFFECT_STATE_CONFIG:
+ break;
+ default:
+ BAD_STATE_ABORT(effect->state, state);
+ }
+ break;
+ case PREPROC_EFFECT_STATE_ACTIVE:
+ switch (effect->state) {
+ case PREPROC_EFFECT_STATE_INIT:
+ case PREPROC_EFFECT_STATE_CREATED:
+ ALOGE("Effect_SetState invalid transition");
+ status = -ENOSYS;
+ break;
+ case PREPROC_EFFECT_STATE_ACTIVE:
+ // enabling an already enabled effect is just ignored
+ break;
+ case PREPROC_EFFECT_STATE_CONFIG:
+ effect->ops->enable(effect);
+ Session_SetProcEnabled(effect->session, effect->procId, true);
+ break;
+ default:
+ BAD_STATE_ABORT(effect->state, state);
+ }
break;
default:
BAD_STATE_ABORT(effect->state, state);
- }
- break;
- case PREPROC_EFFECT_STATE_CREATED:
- switch(effect->state) {
- case PREPROC_EFFECT_STATE_INIT:
- status = effect->ops->create(effect);
- break;
- case PREPROC_EFFECT_STATE_CREATED:
- case PREPROC_EFFECT_STATE_ACTIVE:
- case PREPROC_EFFECT_STATE_CONFIG:
- ALOGE("Effect_SetState invalid transition");
- status = -ENOSYS;
- break;
- default:
- BAD_STATE_ABORT(effect->state, state);
- }
- break;
- case PREPROC_EFFECT_STATE_CONFIG:
- switch(effect->state) {
- case PREPROC_EFFECT_STATE_INIT:
- ALOGE("Effect_SetState invalid transition");
- status = -ENOSYS;
- break;
- case PREPROC_EFFECT_STATE_ACTIVE:
- effect->ops->disable(effect);
- Session_SetProcEnabled(effect->session, effect->procId, false);
- break;
- case PREPROC_EFFECT_STATE_CREATED:
- case PREPROC_EFFECT_STATE_CONFIG:
- break;
- default:
- BAD_STATE_ABORT(effect->state, state);
- }
- break;
- case PREPROC_EFFECT_STATE_ACTIVE:
- switch(effect->state) {
- case PREPROC_EFFECT_STATE_INIT:
- case PREPROC_EFFECT_STATE_CREATED:
- ALOGE("Effect_SetState invalid transition");
- status = -ENOSYS;
- break;
- case PREPROC_EFFECT_STATE_ACTIVE:
- // enabling an already enabled effect is just ignored
- break;
- case PREPROC_EFFECT_STATE_CONFIG:
- effect->ops->enable(effect);
- Session_SetProcEnabled(effect->session, effect->procId, true);
- break;
- default:
- BAD_STATE_ABORT(effect->state, state);
- }
- break;
- default:
- BAD_STATE_ABORT(effect->state, state);
}
if (status == 0) {
effect->state = state;
@@ -1193,8 +1079,7 @@
return status;
}
-int Effect_Init(preproc_effect_t *effect, uint32_t procId)
-{
+int Effect_Init(preproc_effect_t* effect, uint32_t procId) {
if (HasReverseStream(procId)) {
effect->itfe = &sEffectInterfaceReverse;
} else {
@@ -1206,21 +1091,17 @@
return 0;
}
-int Effect_Create(preproc_effect_t *effect,
- preproc_session_t *session,
- effect_handle_t *interface)
-{
+int Effect_Create(preproc_effect_t* effect, preproc_session_t* session,
+ effect_handle_t* interface) {
effect->session = session;
*interface = (effect_handle_t)&effect->itfe;
return Effect_SetState(effect, PREPROC_EFFECT_STATE_CREATED);
}
-int Effect_Release(preproc_effect_t *effect)
-{
+int Effect_Release(preproc_effect_t* effect) {
return Effect_SetState(effect, PREPROC_EFFECT_STATE_INIT);
}
-
//------------------------------------------------------------------------------
// Session functions
//------------------------------------------------------------------------------
@@ -1230,8 +1111,7 @@
static const int kPreprocDefaultSr = 16000;
static const int kPreProcDefaultCnl = 1;
-int Session_Init(preproc_session_t *session)
-{
+int Session_Init(preproc_session_t* session) {
size_t i;
int status = 0;
@@ -1248,11 +1128,8 @@
return status;
}
-
-extern "C" int Session_CreateEffect(preproc_session_t *session,
- int32_t procId,
- effect_handle_t *interface)
-{
+extern "C" int Session_CreateEffect(preproc_session_t* session, int32_t procId,
+ effect_handle_t* interface) {
int status = -ENOMEM;
ALOGV("Session_CreateEffect procId %d, createdMsk %08x", procId, session->createdMsk);
@@ -1265,10 +1142,10 @@
goto error;
}
const webrtc::ProcessingConfig processing_config = {
- {{kPreprocDefaultSr, kPreProcDefaultCnl},
- {kPreprocDefaultSr, kPreProcDefaultCnl},
- {kPreprocDefaultSr, kPreProcDefaultCnl},
- {kPreprocDefaultSr, kPreProcDefaultCnl}}};
+ {{kPreprocDefaultSr, kPreProcDefaultCnl},
+ {kPreprocDefaultSr, kPreProcDefaultCnl},
+ {kPreprocDefaultSr, kPreProcDefaultCnl},
+ {kPreprocDefaultSr, kPreProcDefaultCnl}}};
session->apm->Initialize(processing_config);
session->procFrame = new webrtc::AudioFrame();
if (session->procFrame == NULL) {
@@ -1335,7 +1212,7 @@
goto error;
}
ALOGV("Session_CreateEffect OK");
- session->createdMsk |= (1<<procId);
+ session->createdMsk |= (1 << procId);
return status;
error:
@@ -1346,7 +1223,7 @@
delete session->procFrame;
session->procFrame = NULL;
delete session->apm;
- session->apm = NULL; // NOLINT(clang-analyzer-cplusplus.NewDelete)
+ session->apm = NULL; // NOLINT(clang-analyzer-cplusplus.NewDelete)
#else
delete session->apm;
session->apm = NULL;
@@ -1355,11 +1232,9 @@
return status;
}
-int Session_ReleaseEffect(preproc_session_t *session,
- preproc_effect_t *fx)
-{
+int Session_ReleaseEffect(preproc_session_t* session, preproc_effect_t* fx) {
ALOGW_IF(Effect_Release(fx) != 0, " Effect_Release() failed for proc ID %d", fx->procId);
- session->createdMsk &= ~(1<<fx->procId);
+ session->createdMsk &= ~(1 << fx->procId);
if (session->createdMsk == 0) {
#ifdef WEBRTC_LEGACY
delete session->apm;
@@ -1397,9 +1272,7 @@
return 0;
}
-
-int Session_SetConfig(preproc_session_t *session, effect_config_t *config)
-{
+int Session_SetConfig(preproc_session_t* session, effect_config_t* config) {
uint32_t inCnl = audio_channel_count_from_in_mask(config->inputCfg.channels);
uint32_t outCnl = audio_channel_count_from_in_mask(config->outputCfg.channels);
@@ -1409,8 +1282,8 @@
return -EINVAL;
}
- ALOGV("Session_SetConfig sr %d cnl %08x",
- config->inputCfg.samplingRate, config->inputCfg.channels);
+ ALOGV("Session_SetConfig sr %d cnl %08x", config->inputCfg.samplingRate,
+ config->inputCfg.channels);
#ifdef WEBRTC_LEGACY
int status;
#endif
@@ -1418,8 +1291,7 @@
// AEC implementation is limited to 16kHz
if (config->inputCfg.samplingRate >= 32000 && !(session->createdMsk & (1 << PREPROC_AEC))) {
session->apmSamplingRate = 32000;
- } else
- if (config->inputCfg.samplingRate >= 16000) {
+ } else if (config->inputCfg.samplingRate >= 16000) {
session->apmSamplingRate = 16000;
} else if (config->inputCfg.samplingRate >= 8000) {
session->apmSamplingRate = 8000;
@@ -1427,10 +1299,10 @@
#ifdef WEBRTC_LEGACY
const webrtc::ProcessingConfig processing_config = {
- {{static_cast<int>(session->apmSamplingRate), inCnl},
- {static_cast<int>(session->apmSamplingRate), outCnl},
- {static_cast<int>(session->apmSamplingRate), inCnl},
- {static_cast<int>(session->apmSamplingRate), inCnl}}};
+ {{static_cast<int>(session->apmSamplingRate), inCnl},
+ {static_cast<int>(session->apmSamplingRate), outCnl},
+ {static_cast<int>(session->apmSamplingRate), inCnl},
+ {static_cast<int>(session->apmSamplingRate), inCnl}}};
status = session->apm->Initialize(processing_config);
if (status < 0) {
return -EINVAL;
@@ -1443,11 +1315,11 @@
session->frameCount = session->apmFrameCount;
} else {
#ifdef WEBRTC_LEGACY
- session->frameCount = (session->apmFrameCount * session->samplingRate) /
- session->apmSamplingRate + 1;
+ session->frameCount =
+ (session->apmFrameCount * session->samplingRate) / session->apmSamplingRate + 1;
#else
- session->frameCount = (session->apmFrameCount * session->samplingRate) /
- session->apmSamplingRate;
+ session->frameCount =
+ (session->apmFrameCount * session->samplingRate) / session->apmSamplingRate;
#endif
}
session->inChannelCount = inCnl;
@@ -1477,7 +1349,6 @@
session->framesIn = 0;
session->framesOut = 0;
-
#ifdef WEBRTC_LEGACY
if (session->inResampler != NULL) {
speex_resampler_destroy(session->inResampler);
@@ -1493,36 +1364,30 @@
}
if (session->samplingRate != session->apmSamplingRate) {
int error;
- session->inResampler = speex_resampler_init(session->inChannelCount,
- session->samplingRate,
- session->apmSamplingRate,
- RESAMPLER_QUALITY,
- &error);
+ session->inResampler =
+ speex_resampler_init(session->inChannelCount, session->samplingRate,
+ session->apmSamplingRate, RESAMPLER_QUALITY, &error);
if (session->inResampler == NULL) {
ALOGW("Session_SetConfig Cannot create speex resampler: %s",
- speex_resampler_strerror(error));
+ speex_resampler_strerror(error));
return -EINVAL;
}
- session->outResampler = speex_resampler_init(session->outChannelCount,
- session->apmSamplingRate,
- session->samplingRate,
- RESAMPLER_QUALITY,
- &error);
+ session->outResampler =
+ speex_resampler_init(session->outChannelCount, session->apmSamplingRate,
+ session->samplingRate, RESAMPLER_QUALITY, &error);
if (session->outResampler == NULL) {
ALOGW("Session_SetConfig Cannot create speex resampler: %s",
- speex_resampler_strerror(error));
+ speex_resampler_strerror(error));
speex_resampler_destroy(session->inResampler);
session->inResampler = NULL;
return -EINVAL;
}
- session->revResampler = speex_resampler_init(session->inChannelCount,
- session->samplingRate,
- session->apmSamplingRate,
- RESAMPLER_QUALITY,
- &error);
+ session->revResampler =
+ speex_resampler_init(session->inChannelCount, session->samplingRate,
+ session->apmSamplingRate, RESAMPLER_QUALITY, &error);
if (session->revResampler == NULL) {
ALOGW("Session_SetConfig Cannot create speex resampler: %s",
- speex_resampler_strerror(error));
+ speex_resampler_strerror(error));
speex_resampler_destroy(session->inResampler);
session->inResampler = NULL;
speex_resampler_destroy(session->outResampler);
@@ -1536,8 +1401,7 @@
return 0;
}
-void Session_GetConfig(preproc_session_t *session, effect_config_t *config)
-{
+void Session_GetConfig(preproc_session_t* session, effect_config_t* config) {
memset(config, 0, sizeof(effect_config_t));
config->inputCfg.samplingRate = config->outputCfg.samplingRate = session->samplingRate;
config->inputCfg.format = config->outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
@@ -1548,31 +1412,30 @@
(EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT);
}
-int Session_SetReverseConfig(preproc_session_t *session, effect_config_t *config)
-{
+int Session_SetReverseConfig(preproc_session_t* session, effect_config_t* config) {
if (config->inputCfg.samplingRate != config->outputCfg.samplingRate ||
- config->inputCfg.format != config->outputCfg.format ||
- config->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT) {
+ config->inputCfg.format != config->outputCfg.format ||
+ config->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT) {
return -EINVAL;
}
- ALOGV("Session_SetReverseConfig sr %d cnl %08x",
- config->inputCfg.samplingRate, config->inputCfg.channels);
+ ALOGV("Session_SetReverseConfig sr %d cnl %08x", config->inputCfg.samplingRate,
+ config->inputCfg.channels);
if (session->state < PREPROC_SESSION_STATE_CONFIG) {
return -ENOSYS;
}
if (config->inputCfg.samplingRate != session->samplingRate ||
- config->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT) {
+ config->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT) {
return -EINVAL;
}
uint32_t inCnl = audio_channel_count_from_out_mask(config->inputCfg.channels);
#ifdef WEBRTC_LEGACY
const webrtc::ProcessingConfig processing_config = {
- {{static_cast<int>(session->apmSamplingRate), session->inChannelCount},
- {static_cast<int>(session->apmSamplingRate), session->outChannelCount},
- {static_cast<int>(session->apmSamplingRate), inCnl},
- {static_cast<int>(session->apmSamplingRate), inCnl}}};
+ {{static_cast<int>(session->apmSamplingRate), session->inChannelCount},
+ {static_cast<int>(session->apmSamplingRate), session->outChannelCount},
+ {static_cast<int>(session->apmSamplingRate), inCnl},
+ {static_cast<int>(session->apmSamplingRate), inCnl}}};
int status = session->apm->Initialize(processing_config);
if (status < 0) {
return -EINVAL;
@@ -1590,8 +1453,7 @@
return 0;
}
-void Session_GetReverseConfig(preproc_session_t *session, effect_config_t *config)
-{
+void Session_GetReverseConfig(preproc_session_t* session, effect_config_t* config) {
memset(config, 0, sizeof(effect_config_t));
config->inputCfg.samplingRate = config->outputCfg.samplingRate = session->samplingRate;
config->inputCfg.format = config->outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
@@ -1601,10 +1463,9 @@
(EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT);
}
-void Session_SetProcEnabled(preproc_session_t *session, uint32_t procId, bool enabled)
-{
+void Session_SetProcEnabled(preproc_session_t* session, uint32_t procId, bool enabled) {
if (enabled) {
- if(session->enabledMsk == 0) {
+ if (session->enabledMsk == 0) {
session->framesIn = 0;
#ifdef WEBRTC_LEGACY
if (session->inResampler != NULL) {
@@ -1632,8 +1493,8 @@
session->revEnabledMsk &= ~(1 << procId);
}
}
- ALOGV("Session_SetProcEnabled proc %d, enabled %d enabledMsk %08x revEnabledMsk %08x",
- procId, enabled, session->enabledMsk, session->revEnabledMsk);
+ ALOGV("Session_SetProcEnabled proc %d, enabled %d enabledMsk %08x revEnabledMsk %08x", procId,
+ enabled, session->enabledMsk, session->revEnabledMsk);
session->processedMsk = 0;
if (HasReverseStream(procId)) {
session->revProcessedMsk = 0;
@@ -1647,8 +1508,7 @@
static int sInitStatus = 1;
static preproc_session_t sSessions[PREPROC_NUM_SESSIONS];
-preproc_session_t *PreProc_GetSession(int32_t procId, int32_t sessionId, int32_t ioId)
-{
+preproc_session_t* PreProc_GetSession(int32_t procId, int32_t sessionId, int32_t ioId) {
size_t i;
for (i = 0; i < PREPROC_NUM_SESSIONS; i++) {
if (sSessions[i].id == sessionId) {
@@ -1668,7 +1528,6 @@
return NULL;
}
-
int PreProc_Init() {
size_t i;
int status = 0;
@@ -1683,8 +1542,7 @@
return sInitStatus;
}
-const effect_descriptor_t *PreProc_GetDescriptor(const effect_uuid_t *uuid)
-{
+const effect_descriptor_t* PreProc_GetDescriptor(const effect_uuid_t* uuid) {
size_t i;
for (i = 0; i < PREPROC_NUM_EFFECTS; i++) {
if (memcmp(&sDescriptors[i]->uuid, uuid, sizeof(effect_uuid_t)) == 0) {
@@ -1694,35 +1552,31 @@
return NULL;
}
-
extern "C" {
//------------------------------------------------------------------------------
// Effect Control Interface Implementation
//------------------------------------------------------------------------------
-int PreProcessingFx_Process(effect_handle_t self,
- audio_buffer_t *inBuffer,
- audio_buffer_t *outBuffer)
-{
- preproc_effect_t * effect = (preproc_effect_t *)self;
+int PreProcessingFx_Process(effect_handle_t self, audio_buffer_t* inBuffer,
+ audio_buffer_t* outBuffer) {
+ preproc_effect_t* effect = (preproc_effect_t*)self;
- if (effect == NULL){
+ if (effect == NULL) {
ALOGV("PreProcessingFx_Process() ERROR effect == NULL");
return -EINVAL;
}
- preproc_session_t * session = (preproc_session_t *)effect->session;
+ preproc_session_t* session = (preproc_session_t*)effect->session;
- if (inBuffer == NULL || inBuffer->raw == NULL ||
- outBuffer == NULL || outBuffer->raw == NULL){
+ if (inBuffer == NULL || inBuffer->raw == NULL || outBuffer == NULL || outBuffer->raw == NULL) {
ALOGW("PreProcessingFx_Process() ERROR bad pointer");
return -EINVAL;
}
- session->processedMsk |= (1<<effect->procId);
+ session->processedMsk |= (1 << effect->procId);
-// ALOGV("PreProcessingFx_Process In %d frames enabledMsk %08x processedMsk %08x",
-// inBuffer->frameCount, session->enabledMsk, session->processedMsk);
+ // ALOGV("PreProcessingFx_Process In %d frames enabledMsk %08x processedMsk %08x",
+ // inBuffer->frameCount, session->enabledMsk, session->processedMsk);
if ((session->processedMsk & session->enabledMsk) == session->enabledMsk) {
effect->session->processedMsk = 0;
@@ -1733,11 +1587,9 @@
if (outBuffer->frameCount < fr) {
fr = outBuffer->frameCount;
}
- memcpy(outBuffer->s16,
- session->outBuf,
- fr * session->outChannelCount * sizeof(int16_t));
- memmove(session->outBuf,
- session->outBuf + fr * session->outChannelCount,
+ memcpy(outBuffer->s16, session->outBuf,
+ fr * session->outChannelCount * sizeof(int16_t));
+ memmove(session->outBuf, session->outBuf + fr * session->outChannelCount,
(session->framesOut - fr) * session->outChannelCount * sizeof(int16_t));
session->framesOut -= fr;
framesWr += fr;
@@ -1755,10 +1607,11 @@
fr = inBuffer->frameCount;
}
if (session->inBufSize < session->framesIn + fr) {
- int16_t *buf;
+ int16_t* buf;
session->inBufSize = session->framesIn + fr;
- buf = (int16_t *)realloc(session->inBuf,
- session->inBufSize * session->inChannelCount * sizeof(int16_t));
+ buf = (int16_t*)realloc(
+ session->inBuf,
+ session->inBufSize * session->inChannelCount * sizeof(int16_t));
if (buf == NULL) {
session->framesIn = 0;
free(session->inBuf);
@@ -1767,14 +1620,13 @@
}
session->inBuf = buf;
}
- memcpy(session->inBuf + session->framesIn * session->inChannelCount,
- inBuffer->s16,
+ memcpy(session->inBuf + session->framesIn * session->inChannelCount, inBuffer->s16,
fr * session->inChannelCount * sizeof(int16_t));
#ifdef DUAL_MIC_TEST
pthread_mutex_lock(&gPcmDumpLock);
if (gPcmDumpFh != NULL) {
- fwrite(inBuffer->raw,
- fr * session->inChannelCount * sizeof(int16_t), 1, gPcmDumpFh);
+ fwrite(inBuffer->raw, fr * session->inChannelCount * sizeof(int16_t), 1,
+ gPcmDumpFh);
}
pthread_mutex_unlock(&gPcmDumpLock);
#endif
@@ -1787,21 +1639,13 @@
spx_uint32_t frIn = session->framesIn;
spx_uint32_t frOut = session->apmFrameCount;
if (session->inChannelCount == 1) {
- speex_resampler_process_int(session->inResampler,
- 0,
- session->inBuf,
- &frIn,
- session->procFrame->data_,
- &frOut);
+ speex_resampler_process_int(session->inResampler, 0, session->inBuf, &frIn,
+ session->procFrame->data_, &frOut);
} else {
- speex_resampler_process_interleaved_int(session->inResampler,
- session->inBuf,
- &frIn,
- session->procFrame->data_,
- &frOut);
+ speex_resampler_process_interleaved_int(session->inResampler, session->inBuf, &frIn,
+ session->procFrame->data_, &frOut);
}
- memmove(session->inBuf,
- session->inBuf + frIn * session->inChannelCount,
+ memmove(session->inBuf, session->inBuf + frIn * session->inChannelCount,
(session->framesIn - frIn) * session->inChannelCount * sizeof(int16_t));
session->framesIn -= frIn;
} else {
@@ -1810,14 +1654,13 @@
fr = inBuffer->frameCount;
}
memcpy(session->procFrame->data_ + session->framesIn * session->inChannelCount,
- inBuffer->s16,
- fr * session->inChannelCount * sizeof(int16_t));
+ inBuffer->s16, fr * session->inChannelCount * sizeof(int16_t));
#ifdef DUAL_MIC_TEST
pthread_mutex_lock(&gPcmDumpLock);
if (gPcmDumpFh != NULL) {
- fwrite(inBuffer->raw,
- fr * session->inChannelCount * sizeof(int16_t), 1, gPcmDumpFh);
+ fwrite(inBuffer->raw, fr * session->inChannelCount * sizeof(int16_t), 1,
+ gPcmDumpFh);
}
pthread_mutex_unlock(&gPcmDumpLock);
#endif
@@ -1844,11 +1687,11 @@
}
session->framesIn = 0;
if (int status = effect->session->apm->ProcessStream(
- (const int16_t* const)inBuffer->s16,
- (const webrtc::StreamConfig)effect->session->inputConfig,
- (const webrtc::StreamConfig)effect->session->outputConfig,
- (int16_t* const)outBuffer->s16);
- status != 0) {
+ (const int16_t* const)inBuffer->s16,
+ (const webrtc::StreamConfig)effect->session->inputConfig,
+ (const webrtc::StreamConfig)effect->session->outputConfig,
+ (int16_t* const)outBuffer->s16);
+ status != 0) {
ALOGE("Process Stream failed with error %d\n", status);
return status;
}
@@ -1856,10 +1699,11 @@
#endif
if (session->outBufSize < session->framesOut + session->frameCount) {
- int16_t *buf;
+ int16_t* buf;
session->outBufSize = session->framesOut + session->frameCount;
- buf = (int16_t *)realloc(session->outBuf,
- session->outBufSize * session->outChannelCount * sizeof(int16_t));
+ buf = (int16_t*)realloc(
+ session->outBuf,
+ session->outBufSize * session->outChannelCount * sizeof(int16_t));
if (buf == NULL) {
session->framesOut = 0;
free(session->outBuf);
@@ -1874,18 +1718,13 @@
spx_uint32_t frIn = session->apmFrameCount;
spx_uint32_t frOut = session->frameCount;
if (session->inChannelCount == 1) {
- speex_resampler_process_int(session->outResampler,
- 0,
- session->procFrame->data_,
- &frIn,
- session->outBuf + session->framesOut * session->outChannelCount,
- &frOut);
+ speex_resampler_process_int(
+ session->outResampler, 0, session->procFrame->data_, &frIn,
+ session->outBuf + session->framesOut * session->outChannelCount, &frOut);
} else {
- speex_resampler_process_interleaved_int(session->outResampler,
- session->procFrame->data_,
- &frIn,
- session->outBuf + session->framesOut * session->outChannelCount,
- &frOut);
+ speex_resampler_process_interleaved_int(
+ session->outResampler, session->procFrame->data_, &frIn,
+ session->outBuf + session->framesOut * session->outChannelCount, &frOut);
}
session->framesOut += frOut;
} else {
@@ -1901,11 +1740,9 @@
if (framesRq - framesWr < fr) {
fr = framesRq - framesWr;
}
- memcpy(outBuffer->s16 + framesWr * session->outChannelCount,
- session->outBuf,
- fr * session->outChannelCount * sizeof(int16_t));
- memmove(session->outBuf,
- session->outBuf + fr * session->outChannelCount,
+ memcpy(outBuffer->s16 + framesWr * session->outChannelCount, session->outBuf,
+ fr * session->outChannelCount * sizeof(int16_t));
+ memmove(session->outBuf, session->outBuf + fr * session->outChannelCount,
(session->framesOut - fr) * session->outChannelCount * sizeof(int16_t));
session->framesOut -= fr;
outBuffer->frameCount += fr;
@@ -1916,39 +1753,32 @@
}
}
-int PreProcessingFx_Command(effect_handle_t self,
- uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t *replySize,
- void *pReplyData)
-{
- preproc_effect_t * effect = (preproc_effect_t *) self;
+int PreProcessingFx_Command(effect_handle_t self, uint32_t cmdCode, uint32_t cmdSize,
+ void* pCmdData, uint32_t* replySize, void* pReplyData) {
+ preproc_effect_t* effect = (preproc_effect_t*)self;
- if (effect == NULL){
+ if (effect == NULL) {
return -EINVAL;
}
- //ALOGV("PreProcessingFx_Command: command %d cmdSize %d",cmdCode, cmdSize);
+ // ALOGV("PreProcessingFx_Command: command %d cmdSize %d",cmdCode, cmdSize);
- switch (cmdCode){
+ switch (cmdCode) {
case EFFECT_CMD_INIT:
- if (pReplyData == NULL || *replySize != sizeof(int)){
+ if (pReplyData == NULL || *replySize != sizeof(int)) {
return -EINVAL;
}
if (effect->ops->init) {
effect->ops->init(effect);
}
- *(int *)pReplyData = 0;
+ *(int*)pReplyData = 0;
break;
case EFFECT_CMD_SET_CONFIG: {
- if (pCmdData == NULL||
- cmdSize != sizeof(effect_config_t)||
- pReplyData == NULL||
- *replySize != sizeof(int)){
+ if (pCmdData == NULL || cmdSize != sizeof(effect_config_t) || pReplyData == NULL ||
+ *replySize != sizeof(int)) {
ALOGV("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_SET_CONFIG: ERROR");
+ "EFFECT_CMD_SET_CONFIG: ERROR");
return -EINVAL;
}
#ifdef DUAL_MIC_TEST
@@ -1959,55 +1789,51 @@
effect->session->enabledMsk = 0;
}
#endif
- *(int *)pReplyData = Session_SetConfig(effect->session, (effect_config_t *)pCmdData);
+ *(int*)pReplyData = Session_SetConfig(effect->session, (effect_config_t*)pCmdData);
#ifdef DUAL_MIC_TEST
if (gDualMicEnabled) {
effect->session->enabledMsk = enabledMsk;
}
#endif
- if (*(int *)pReplyData != 0) {
+ if (*(int*)pReplyData != 0) {
break;
}
if (effect->state != PREPROC_EFFECT_STATE_ACTIVE) {
- *(int *)pReplyData = Effect_SetState(effect, PREPROC_EFFECT_STATE_CONFIG);
+ *(int*)pReplyData = Effect_SetState(effect, PREPROC_EFFECT_STATE_CONFIG);
}
- } break;
+ } break;
case EFFECT_CMD_GET_CONFIG:
- if (pReplyData == NULL ||
- *replySize != sizeof(effect_config_t)) {
+ if (pReplyData == NULL || *replySize != sizeof(effect_config_t)) {
ALOGV("\tLVM_ERROR : PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_GET_CONFIG: ERROR");
+ "EFFECT_CMD_GET_CONFIG: ERROR");
return -EINVAL;
}
- Session_GetConfig(effect->session, (effect_config_t *)pReplyData);
+ Session_GetConfig(effect->session, (effect_config_t*)pReplyData);
break;
case EFFECT_CMD_SET_CONFIG_REVERSE:
- if (pCmdData == NULL ||
- cmdSize != sizeof(effect_config_t) ||
- pReplyData == NULL ||
+ if (pCmdData == NULL || cmdSize != sizeof(effect_config_t) || pReplyData == NULL ||
*replySize != sizeof(int)) {
ALOGV("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_SET_CONFIG_REVERSE: ERROR");
+ "EFFECT_CMD_SET_CONFIG_REVERSE: ERROR");
return -EINVAL;
}
- *(int *)pReplyData = Session_SetReverseConfig(effect->session,
- (effect_config_t *)pCmdData);
- if (*(int *)pReplyData != 0) {
+ *(int*)pReplyData =
+ Session_SetReverseConfig(effect->session, (effect_config_t*)pCmdData);
+ if (*(int*)pReplyData != 0) {
break;
}
break;
case EFFECT_CMD_GET_CONFIG_REVERSE:
- if (pReplyData == NULL ||
- *replySize != sizeof(effect_config_t)){
+ if (pReplyData == NULL || *replySize != sizeof(effect_config_t)) {
ALOGV("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_GET_CONFIG_REVERSE: ERROR");
+ "EFFECT_CMD_GET_CONFIG_REVERSE: ERROR");
return -EINVAL;
}
- Session_GetReverseConfig(effect->session, (effect_config_t *)pCmdData);
+ Session_GetReverseConfig(effect->session, (effect_config_t*)pCmdData);
break;
case EFFECT_CMD_RESET:
@@ -2017,80 +1843,74 @@
break;
case EFFECT_CMD_GET_PARAM: {
- effect_param_t *p = (effect_param_t *)pCmdData;
+ effect_param_t* p = (effect_param_t*)pCmdData;
if (pCmdData == NULL || cmdSize < sizeof(effect_param_t) ||
- cmdSize < (sizeof(effect_param_t) + p->psize) ||
- pReplyData == NULL || replySize == NULL ||
- *replySize < (sizeof(effect_param_t) + p->psize)){
+ cmdSize < (sizeof(effect_param_t) + p->psize) || pReplyData == NULL ||
+ replySize == NULL || *replySize < (sizeof(effect_param_t) + p->psize)) {
ALOGV("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_GET_PARAM: ERROR");
+ "EFFECT_CMD_GET_PARAM: ERROR");
return -EINVAL;
}
memcpy(pReplyData, pCmdData, sizeof(effect_param_t) + p->psize);
- p = (effect_param_t *)pReplyData;
+ p = (effect_param_t*)pReplyData;
int voffset = ((p->psize - 1) / sizeof(int32_t) + 1) * sizeof(int32_t);
if (effect->ops->get_parameter) {
- p->status = effect->ops->get_parameter(effect, p->data,
- &p->vsize,
- p->data + voffset);
+ p->status =
+ effect->ops->get_parameter(effect, p->data, &p->vsize, p->data + voffset);
*replySize = sizeof(effect_param_t) + voffset + p->vsize;
}
} break;
- case EFFECT_CMD_SET_PARAM:{
- if (pCmdData == NULL||
- cmdSize < sizeof(effect_param_t) ||
- pReplyData == NULL || replySize == NULL ||
- *replySize != sizeof(int32_t)){
+ case EFFECT_CMD_SET_PARAM: {
+ if (pCmdData == NULL || cmdSize < sizeof(effect_param_t) || pReplyData == NULL ||
+ replySize == NULL || *replySize != sizeof(int32_t)) {
ALOGV("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_SET_PARAM: ERROR");
+ "EFFECT_CMD_SET_PARAM: ERROR");
return -EINVAL;
}
- effect_param_t *p = (effect_param_t *) pCmdData;
+ effect_param_t* p = (effect_param_t*)pCmdData;
- if (p->psize != sizeof(int32_t)){
+ if (p->psize != sizeof(int32_t)) {
ALOGV("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_SET_PARAM: ERROR, psize is not sizeof(int32_t)");
+ "EFFECT_CMD_SET_PARAM: ERROR, psize is not sizeof(int32_t)");
return -EINVAL;
}
if (effect->ops->set_parameter) {
- *(int *)pReplyData = effect->ops->set_parameter(effect,
- (void *)p->data,
- p->data + p->psize);
+ *(int*)pReplyData =
+ effect->ops->set_parameter(effect, (void*)p->data, p->data + p->psize);
}
} break;
case EFFECT_CMD_ENABLE:
- if (pReplyData == NULL || replySize == NULL || *replySize != sizeof(int)){
+ if (pReplyData == NULL || replySize == NULL || *replySize != sizeof(int)) {
ALOGV("PreProcessingFx_Command cmdCode Case: EFFECT_CMD_ENABLE: ERROR");
return -EINVAL;
}
- *(int *)pReplyData = Effect_SetState(effect, PREPROC_EFFECT_STATE_ACTIVE);
+ *(int*)pReplyData = Effect_SetState(effect, PREPROC_EFFECT_STATE_ACTIVE);
break;
case EFFECT_CMD_DISABLE:
- if (pReplyData == NULL || replySize == NULL || *replySize != sizeof(int)){
+ if (pReplyData == NULL || replySize == NULL || *replySize != sizeof(int)) {
ALOGV("PreProcessingFx_Command cmdCode Case: EFFECT_CMD_DISABLE: ERROR");
return -EINVAL;
}
- *(int *)pReplyData = Effect_SetState(effect, PREPROC_EFFECT_STATE_CONFIG);
+ *(int*)pReplyData = Effect_SetState(effect, PREPROC_EFFECT_STATE_CONFIG);
break;
case EFFECT_CMD_SET_DEVICE:
case EFFECT_CMD_SET_INPUT_DEVICE:
- if (pCmdData == NULL ||
- cmdSize != sizeof(uint32_t)) {
+ if (pCmdData == NULL || cmdSize != sizeof(uint32_t)) {
ALOGV("PreProcessingFx_Command cmdCode Case: EFFECT_CMD_SET_DEVICE: ERROR");
return -EINVAL;
}
if (effect->ops->set_device) {
- effect->ops->set_device(effect, *(uint32_t *)pCmdData);
+ effect->ops->set_device(effect, *(uint32_t*)pCmdData);
}
break;
@@ -2101,30 +1921,30 @@
#ifdef DUAL_MIC_TEST
///// test commands start
case PREPROC_CMD_DUAL_MIC_ENABLE: {
- if (pCmdData == NULL|| cmdSize != sizeof(uint32_t) ||
- pReplyData == NULL || replySize == NULL) {
+ if (pCmdData == NULL || cmdSize != sizeof(uint32_t) || pReplyData == NULL ||
+ replySize == NULL) {
ALOGE("PreProcessingFx_Command cmdCode Case: "
- "PREPROC_CMD_DUAL_MIC_ENABLE: ERROR");
+ "PREPROC_CMD_DUAL_MIC_ENABLE: ERROR");
*replySize = 0;
return -EINVAL;
}
- gDualMicEnabled = *(bool *)pCmdData;
+ gDualMicEnabled = *(bool*)pCmdData;
if (gDualMicEnabled) {
effect->aux_channels_on = sHasAuxChannels[effect->procId];
} else {
effect->aux_channels_on = false;
}
- effect->cur_channel_config = (effect->session->inChannelCount == 1) ?
- CHANNEL_CFG_MONO : CHANNEL_CFG_STEREO;
+ effect->cur_channel_config =
+ (effect->session->inChannelCount == 1) ? CHANNEL_CFG_MONO : CHANNEL_CFG_STEREO;
ALOGV("PREPROC_CMD_DUAL_MIC_ENABLE: %s", gDualMicEnabled ? "enabled" : "disabled");
*replySize = sizeof(int);
- *(int *)pReplyData = 0;
- } break;
+ *(int*)pReplyData = 0;
+ } break;
case PREPROC_CMD_DUAL_MIC_PCM_DUMP_START: {
- if (pCmdData == NULL|| pReplyData == NULL || replySize == NULL) {
+ if (pCmdData == NULL || pReplyData == NULL || replySize == NULL) {
ALOGE("PreProcessingFx_Command cmdCode Case: "
- "PREPROC_CMD_DUAL_MIC_PCM_DUMP_START: ERROR");
+ "PREPROC_CMD_DUAL_MIC_PCM_DUMP_START: ERROR");
*replySize = 0;
return -EINVAL;
}
@@ -2133,20 +1953,19 @@
fclose(gPcmDumpFh);
gPcmDumpFh = NULL;
}
- char *path = strndup((char *)pCmdData, cmdSize);
- gPcmDumpFh = fopen((char *)path, "wb");
+ char* path = strndup((char*)pCmdData, cmdSize);
+ gPcmDumpFh = fopen((char*)path, "wb");
pthread_mutex_unlock(&gPcmDumpLock);
- ALOGV("PREPROC_CMD_DUAL_MIC_PCM_DUMP_START: path %s gPcmDumpFh %p",
- path, gPcmDumpFh);
+ ALOGV("PREPROC_CMD_DUAL_MIC_PCM_DUMP_START: path %s gPcmDumpFh %p", path, gPcmDumpFh);
ALOGE_IF(gPcmDumpFh <= 0, "gPcmDumpFh open error %d %s", errno, strerror(errno));
free(path);
*replySize = sizeof(int);
- *(int *)pReplyData = 0;
- } break;
+ *(int*)pReplyData = 0;
+ } break;
case PREPROC_CMD_DUAL_MIC_PCM_DUMP_STOP: {
if (pReplyData == NULL || replySize == NULL) {
ALOGE("PreProcessingFx_Command cmdCode Case: "
- "PREPROC_CMD_DUAL_MIC_PCM_DUMP_STOP: ERROR");
+ "PREPROC_CMD_DUAL_MIC_PCM_DUMP_STOP: ERROR");
*replySize = 0;
return -EINVAL;
}
@@ -2158,118 +1977,116 @@
pthread_mutex_unlock(&gPcmDumpLock);
ALOGV("PREPROC_CMD_DUAL_MIC_PCM_DUMP_STOP");
*replySize = sizeof(int);
- *(int *)pReplyData = 0;
- } break;
- ///// test commands end
+ *(int*)pReplyData = 0;
+ } break;
+ ///// test commands end
case EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS: {
- if(!gDualMicEnabled) {
+ if (!gDualMicEnabled) {
return -EINVAL;
}
- if (pCmdData == NULL|| cmdSize != 2 * sizeof(uint32_t) ||
- pReplyData == NULL || replySize == NULL) {
+ if (pCmdData == NULL || cmdSize != 2 * sizeof(uint32_t) || pReplyData == NULL ||
+ replySize == NULL) {
ALOGE("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS: ERROR");
+ "EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS: ERROR");
*replySize = 0;
return -EINVAL;
}
- if (*(uint32_t *)pCmdData != EFFECT_FEATURE_AUX_CHANNELS ||
- !effect->aux_channels_on) {
+ if (*(uint32_t*)pCmdData != EFFECT_FEATURE_AUX_CHANNELS || !effect->aux_channels_on) {
ALOGV("PreProcessingFx_Command feature EFFECT_FEATURE_AUX_CHANNELS not supported by"
- " fx %d", effect->procId);
- *(uint32_t *)pReplyData = -ENOSYS;
+ " fx %d",
+ effect->procId);
+ *(uint32_t*)pReplyData = -ENOSYS;
*replySize = sizeof(uint32_t);
break;
}
- size_t num_configs = *((uint32_t *)pCmdData + 1);
- if (*replySize < (2 * sizeof(uint32_t) +
- num_configs * sizeof(channel_config_t))) {
+ size_t num_configs = *((uint32_t*)pCmdData + 1);
+ if (*replySize < (2 * sizeof(uint32_t) + num_configs * sizeof(channel_config_t))) {
*replySize = 0;
return -EINVAL;
}
- *((uint32_t *)pReplyData + 1) = CHANNEL_CFG_CNT;
+ *((uint32_t*)pReplyData + 1) = CHANNEL_CFG_CNT;
if (num_configs < CHANNEL_CFG_CNT ||
- *replySize < (2 * sizeof(uint32_t) +
- CHANNEL_CFG_CNT * sizeof(channel_config_t))) {
- *(uint32_t *)pReplyData = -ENOMEM;
+ *replySize < (2 * sizeof(uint32_t) + CHANNEL_CFG_CNT * sizeof(channel_config_t))) {
+ *(uint32_t*)pReplyData = -ENOMEM;
} else {
num_configs = CHANNEL_CFG_CNT;
- *(uint32_t *)pReplyData = 0;
+ *(uint32_t*)pReplyData = 0;
}
ALOGV("PreProcessingFx_Command EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS num config %d",
num_configs);
*replySize = 2 * sizeof(uint32_t) + num_configs * sizeof(channel_config_t);
- *((uint32_t *)pReplyData + 1) = num_configs;
- memcpy((uint32_t *)pReplyData + 2, &sDualMicConfigs, num_configs * sizeof(channel_config_t));
- } break;
+ *((uint32_t*)pReplyData + 1) = num_configs;
+ memcpy((uint32_t*)pReplyData + 2, &sDualMicConfigs,
+ num_configs * sizeof(channel_config_t));
+ } break;
case EFFECT_CMD_GET_FEATURE_CONFIG:
- if(!gDualMicEnabled) {
+ if (!gDualMicEnabled) {
return -EINVAL;
}
- if (pCmdData == NULL|| cmdSize != sizeof(uint32_t) ||
- pReplyData == NULL || replySize == NULL ||
- *replySize < sizeof(uint32_t) + sizeof(channel_config_t)) {
+ if (pCmdData == NULL || cmdSize != sizeof(uint32_t) || pReplyData == NULL ||
+ replySize == NULL || *replySize < sizeof(uint32_t) + sizeof(channel_config_t)) {
ALOGE("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_GET_FEATURE_CONFIG: ERROR");
+ "EFFECT_CMD_GET_FEATURE_CONFIG: ERROR");
return -EINVAL;
}
- if (*(uint32_t *)pCmdData != EFFECT_FEATURE_AUX_CHANNELS || !effect->aux_channels_on) {
- *(uint32_t *)pReplyData = -ENOSYS;
+ if (*(uint32_t*)pCmdData != EFFECT_FEATURE_AUX_CHANNELS || !effect->aux_channels_on) {
+ *(uint32_t*)pReplyData = -ENOSYS;
*replySize = sizeof(uint32_t);
break;
}
ALOGV("PreProcessingFx_Command EFFECT_CMD_GET_FEATURE_CONFIG");
- *(uint32_t *)pReplyData = 0;
+ *(uint32_t*)pReplyData = 0;
*replySize = sizeof(uint32_t) + sizeof(channel_config_t);
- memcpy((uint32_t *)pReplyData + 1,
- &sDualMicConfigs[effect->cur_channel_config],
+ memcpy((uint32_t*)pReplyData + 1, &sDualMicConfigs[effect->cur_channel_config],
sizeof(channel_config_t));
break;
case EFFECT_CMD_SET_FEATURE_CONFIG: {
ALOGV("PreProcessingFx_Command EFFECT_CMD_SET_FEATURE_CONFIG: "
- "gDualMicEnabled %d effect->aux_channels_on %d",
+ "gDualMicEnabled %d effect->aux_channels_on %d",
gDualMicEnabled, effect->aux_channels_on);
- if(!gDualMicEnabled) {
+ if (!gDualMicEnabled) {
return -EINVAL;
}
- if (pCmdData == NULL|| cmdSize != (sizeof(uint32_t) + sizeof(channel_config_t)) ||
- pReplyData == NULL || replySize == NULL ||
- *replySize < sizeof(uint32_t)) {
+ if (pCmdData == NULL || cmdSize != (sizeof(uint32_t) + sizeof(channel_config_t)) ||
+ pReplyData == NULL || replySize == NULL || *replySize < sizeof(uint32_t)) {
ALOGE("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_SET_FEATURE_CONFIG: ERROR\n"
- "pCmdData %p cmdSize %d pReplyData %p replySize %p *replySize %d",
- pCmdData, cmdSize, pReplyData, replySize, replySize ? *replySize : -1);
+ "EFFECT_CMD_SET_FEATURE_CONFIG: ERROR\n"
+ "pCmdData %p cmdSize %d pReplyData %p replySize %p *replySize %d",
+ pCmdData, cmdSize, pReplyData, replySize, replySize ? *replySize : -1);
return -EINVAL;
}
*replySize = sizeof(uint32_t);
- if (*(uint32_t *)pCmdData != EFFECT_FEATURE_AUX_CHANNELS || !effect->aux_channels_on) {
- *(uint32_t *)pReplyData = -ENOSYS;
+ if (*(uint32_t*)pCmdData != EFFECT_FEATURE_AUX_CHANNELS || !effect->aux_channels_on) {
+ *(uint32_t*)pReplyData = -ENOSYS;
ALOGV("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_SET_FEATURE_CONFIG: ERROR\n"
- "CmdData %d effect->aux_channels_on %d",
- *(uint32_t *)pCmdData, effect->aux_channels_on);
+ "EFFECT_CMD_SET_FEATURE_CONFIG: ERROR\n"
+ "CmdData %d effect->aux_channels_on %d",
+ *(uint32_t*)pCmdData, effect->aux_channels_on);
break;
}
size_t i;
- for (i = 0; i < CHANNEL_CFG_CNT;i++) {
- if (memcmp((uint32_t *)pCmdData + 1,
- &sDualMicConfigs[i], sizeof(channel_config_t)) == 0) {
+ for (i = 0; i < CHANNEL_CFG_CNT; i++) {
+ if (memcmp((uint32_t*)pCmdData + 1, &sDualMicConfigs[i],
+ sizeof(channel_config_t)) == 0) {
break;
}
}
if (i == CHANNEL_CFG_CNT) {
- *(uint32_t *)pReplyData = -EINVAL;
+ *(uint32_t*)pReplyData = -EINVAL;
ALOGW("PreProcessingFx_Command EFFECT_CMD_SET_FEATURE_CONFIG invalid config"
- "[%08x].[%08x]", *((uint32_t *)pCmdData + 1), *((uint32_t *)pCmdData + 2));
+ "[%08x].[%08x]",
+ *((uint32_t*)pCmdData + 1), *((uint32_t*)pCmdData + 2));
} else {
effect->cur_channel_config = i;
- *(uint32_t *)pReplyData = 0;
+ *(uint32_t*)pReplyData = 0;
ALOGV("PreProcessingFx_Command EFFECT_CMD_SET_FEATURE_CONFIG New config"
- "[%08x].[%08x]", sDualMicConfigs[i].main_channels, sDualMicConfigs[i].aux_channels);
+ "[%08x].[%08x]",
+ sDualMicConfigs[i].main_channels, sDualMicConfigs[i].aux_channels);
}
- } break;
+ } break;
#endif
default:
return -EINVAL;
@@ -2277,11 +2094,8 @@
return 0;
}
-
-int PreProcessingFx_GetDescriptor(effect_handle_t self,
- effect_descriptor_t *pDescriptor)
-{
- preproc_effect_t * effect = (preproc_effect_t *) self;
+int PreProcessingFx_GetDescriptor(effect_handle_t self, effect_descriptor_t* pDescriptor) {
+ preproc_effect_t* effect = (preproc_effect_t*)self;
if (effect == NULL || pDescriptor == NULL) {
return -EINVAL;
@@ -2292,28 +2106,26 @@
return 0;
}
-int PreProcessingFx_ProcessReverse(effect_handle_t self,
- audio_buffer_t *inBuffer,
- audio_buffer_t *outBuffer __unused)
-{
- preproc_effect_t * effect = (preproc_effect_t *)self;
+int PreProcessingFx_ProcessReverse(effect_handle_t self, audio_buffer_t* inBuffer,
+ audio_buffer_t* outBuffer __unused) {
+ preproc_effect_t* effect = (preproc_effect_t*)self;
- if (effect == NULL){
+ if (effect == NULL) {
ALOGW("PreProcessingFx_ProcessReverse() ERROR effect == NULL");
return -EINVAL;
}
- preproc_session_t * session = (preproc_session_t *)effect->session;
+ preproc_session_t* session = (preproc_session_t*)effect->session;
- if (inBuffer == NULL || inBuffer->raw == NULL){
+ if (inBuffer == NULL || inBuffer->raw == NULL) {
ALOGW("PreProcessingFx_ProcessReverse() ERROR bad pointer");
return -EINVAL;
}
- session->revProcessedMsk |= (1<<effect->procId);
+ session->revProcessedMsk |= (1 << effect->procId);
-// ALOGV("PreProcessingFx_ProcessReverse In %d frames revEnabledMsk %08x revProcessedMsk %08x",
-// inBuffer->frameCount, session->revEnabledMsk, session->revProcessedMsk);
-
+ // ALOGV("PreProcessingFx_ProcessReverse In %d frames revEnabledMsk %08x revProcessedMsk
+ // %08x",
+ // inBuffer->frameCount, session->revEnabledMsk, session->revProcessedMsk);
if ((session->revProcessedMsk & session->revEnabledMsk) == session->revEnabledMsk) {
effect->session->revProcessedMsk = 0;
@@ -2324,10 +2136,11 @@
fr = inBuffer->frameCount;
}
if (session->revBufSize < session->framesRev + fr) {
- int16_t *buf;
+ int16_t* buf;
session->revBufSize = session->framesRev + fr;
- buf = (int16_t *)realloc(session->revBuf,
- session->revBufSize * session->inChannelCount * sizeof(int16_t));
+ buf = (int16_t*)realloc(
+ session->revBuf,
+ session->revBufSize * session->inChannelCount * sizeof(int16_t));
if (buf == NULL) {
session->framesRev = 0;
free(session->revBuf);
@@ -2336,8 +2149,7 @@
}
session->revBuf = buf;
}
- memcpy(session->revBuf + session->framesRev * session->inChannelCount,
- inBuffer->s16,
+ memcpy(session->revBuf + session->framesRev * session->inChannelCount, inBuffer->s16,
fr * session->inChannelCount * sizeof(int16_t));
session->framesRev += fr;
@@ -2348,21 +2160,13 @@
spx_uint32_t frIn = session->framesRev;
spx_uint32_t frOut = session->apmFrameCount;
if (session->inChannelCount == 1) {
- speex_resampler_process_int(session->revResampler,
- 0,
- session->revBuf,
- &frIn,
- session->revFrame->data_,
- &frOut);
+ speex_resampler_process_int(session->revResampler, 0, session->revBuf, &frIn,
+ session->revFrame->data_, &frOut);
} else {
- speex_resampler_process_interleaved_int(session->revResampler,
- session->revBuf,
- &frIn,
- session->revFrame->data_,
- &frOut);
+ speex_resampler_process_interleaved_int(session->revResampler, session->revBuf,
+ &frIn, session->revFrame->data_, &frOut);
}
- memmove(session->revBuf,
- session->revBuf + frIn * session->inChannelCount,
+ memmove(session->revBuf, session->revBuf + frIn * session->inChannelCount,
(session->framesRev - frIn) * session->inChannelCount * sizeof(int16_t));
session->framesRev -= frIn;
} else {
@@ -2371,8 +2175,7 @@
fr = inBuffer->frameCount;
}
memcpy(session->revFrame->data_ + session->framesRev * session->inChannelCount,
- inBuffer->s16,
- fr * session->inChannelCount * sizeof(int16_t));
+ inBuffer->s16, fr * session->inChannelCount * sizeof(int16_t));
session->framesRev += fr;
inBuffer->frameCount = fr;
if (session->framesRev < session->frameCount) {
@@ -2394,11 +2197,11 @@
}
session->framesRev = 0;
if (int status = effect->session->apm->ProcessReverseStream(
- (const int16_t* const)inBuffer->s16,
- (const webrtc::StreamConfig)effect->session->revConfig,
- (const webrtc::StreamConfig)effect->session->revConfig,
- (int16_t* const)outBuffer->s16);
- status != 0) {
+ (const int16_t* const)inBuffer->s16,
+ (const webrtc::StreamConfig)effect->session->revConfig,
+ (const webrtc::StreamConfig)effect->session->revConfig,
+ (int16_t* const)outBuffer->s16);
+ status != 0) {
ALOGE("Process Reverse Stream failed with error %d\n", status);
return status;
}
@@ -2409,42 +2212,31 @@
}
}
-
// effect_handle_t interface implementation for effect
const struct effect_interface_s sEffectInterface = {
- PreProcessingFx_Process,
- PreProcessingFx_Command,
- PreProcessingFx_GetDescriptor,
- NULL
-};
+ PreProcessingFx_Process, PreProcessingFx_Command, PreProcessingFx_GetDescriptor, NULL};
const struct effect_interface_s sEffectInterfaceReverse = {
- PreProcessingFx_Process,
- PreProcessingFx_Command,
- PreProcessingFx_GetDescriptor,
- PreProcessingFx_ProcessReverse
-};
+ PreProcessingFx_Process, PreProcessingFx_Command, PreProcessingFx_GetDescriptor,
+ PreProcessingFx_ProcessReverse};
//------------------------------------------------------------------------------
// Effect Library Interface Implementation
//------------------------------------------------------------------------------
-int PreProcessingLib_Create(const effect_uuid_t *uuid,
- int32_t sessionId,
- int32_t ioId,
- effect_handle_t *pInterface)
-{
+int PreProcessingLib_Create(const effect_uuid_t* uuid, int32_t sessionId, int32_t ioId,
+ effect_handle_t* pInterface) {
ALOGV("EffectCreate: uuid: %08x session %d IO: %d", uuid->timeLow, sessionId, ioId);
int status;
- const effect_descriptor_t *desc;
- preproc_session_t *session;
+ const effect_descriptor_t* desc;
+ preproc_session_t* session;
uint32_t procId;
if (PreProc_Init() != 0) {
return sInitStatus;
}
- desc = PreProc_GetDescriptor(uuid);
+ desc = PreProc_GetDescriptor(uuid);
if (desc == NULL) {
ALOGW("EffectCreate: fx not found uuid: %08x", uuid->timeLow);
return -EINVAL;
@@ -2465,14 +2257,13 @@
return status;
}
-int PreProcessingLib_Release(effect_handle_t interface)
-{
+int PreProcessingLib_Release(effect_handle_t interface) {
ALOGV("EffectRelease start %p", interface);
if (PreProc_Init() != 0) {
return sInitStatus;
}
- preproc_effect_t *fx = (preproc_effect_t *)interface;
+ preproc_effect_t* fx = (preproc_effect_t*)interface;
if (fx->session->id == 0) {
return -EINVAL;
@@ -2480,17 +2271,15 @@
return Session_ReleaseEffect(fx->session, fx);
}
-int PreProcessingLib_GetDescriptor(const effect_uuid_t *uuid,
- effect_descriptor_t *pDescriptor) {
-
- if (pDescriptor == NULL || uuid == NULL){
+int PreProcessingLib_GetDescriptor(const effect_uuid_t* uuid, effect_descriptor_t* pDescriptor) {
+ if (pDescriptor == NULL || uuid == NULL) {
return -EINVAL;
}
- const effect_descriptor_t *desc = PreProc_GetDescriptor(uuid);
+ const effect_descriptor_t* desc = PreProc_GetDescriptor(uuid);
if (desc == NULL) {
ALOGV("PreProcessingLib_GetDescriptor() not found");
- return -EINVAL;
+ return -EINVAL;
}
ALOGV("PreProcessingLib_GetDescriptor() got fx %s", desc->name);
@@ -2500,15 +2289,13 @@
}
// This is the only symbol that needs to be exported
-__attribute__ ((visibility ("default")))
-audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
- .tag = AUDIO_EFFECT_LIBRARY_TAG,
- .version = EFFECT_LIBRARY_API_VERSION,
- .name = "Audio Preprocessing Library",
- .implementor = "The Android Open Source Project",
- .create_effect = PreProcessingLib_Create,
- .release_effect = PreProcessingLib_Release,
- .get_descriptor = PreProcessingLib_GetDescriptor
-};
+__attribute__((visibility("default"))) audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
+ .tag = AUDIO_EFFECT_LIBRARY_TAG,
+ .version = EFFECT_LIBRARY_API_VERSION,
+ .name = "Audio Preprocessing Library",
+ .implementor = "The Android Open Source Project",
+ .create_effect = PreProcessingLib_Create,
+ .release_effect = PreProcessingLib_Release,
+ .get_descriptor = PreProcessingLib_GetDescriptor};
-}; // extern "C"
+}; // extern "C"
diff --git a/media/libeffects/preprocessing/benchmarks/Android.bp b/media/libeffects/preprocessing/benchmarks/Android.bp
new file mode 100644
index 0000000..2808293
--- /dev/null
+++ b/media/libeffects/preprocessing/benchmarks/Android.bp
@@ -0,0 +1,51 @@
+cc_benchmark {
+ name: "preprocessing_legacy_benchmark",
+ vendor: true,
+ relative_install_path: "soundfx",
+ srcs: ["preprocessing_benchmark.cpp"],
+ shared_libs: [
+ "libaudiopreprocessing_legacy",
+ "libaudioutils",
+ "liblog",
+ "libutils",
+ "libwebrtc_audio_preprocessing",
+ ],
+ cflags: [
+ "-DWEBRTC_POSIX",
+ "-DWEBRTC_LEGACY",
+ "-fvisibility=default",
+ "-Wall",
+ "-Werror",
+ "-Wextra",
+ ],
+ header_libs: [
+ "libaudioeffects",
+ "libhardware_headers",
+ "libwebrtc_absl_headers",
+ ],
+}
+
+cc_benchmark {
+ name: "preprocessing_benchmark",
+ vendor: true,
+ relative_install_path: "soundfx",
+ srcs: ["preprocessing_benchmark.cpp"],
+ shared_libs: [
+ "libaudiopreprocessing",
+ "libaudioutils",
+ "liblog",
+ "libutils",
+ ],
+ cflags: [
+ "-DWEBRTC_POSIX",
+ "-fvisibility=default",
+ "-Wall",
+ "-Werror",
+ "-Wextra",
+ ],
+ header_libs: [
+ "libaudioeffects",
+ "libhardware_headers",
+ "libwebrtc_absl_headers",
+ ],
+}
diff --git a/media/libeffects/preprocessing/benchmarks/preprocessing_benchmark.cpp b/media/libeffects/preprocessing/benchmarks/preprocessing_benchmark.cpp
new file mode 100644
index 0000000..3a0ad6d
--- /dev/null
+++ b/media/libeffects/preprocessing/benchmarks/preprocessing_benchmark.cpp
@@ -0,0 +1,246 @@
+/*
+ * Copyright 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/*******************************************************************
+ * A test result running on Pixel 3 for comparison.
+ * The first parameter indicates the channel mask index.
+ * The second parameter indicates the effect index.
+ * 0: Automatic Gain Control,
+ * 1: Acoustic Echo Canceler,
+ * 2: Noise Suppressor,
+ * 3: Automatic Gain Control 2
+ * ---------------------------------------------------------------
+ * Benchmark Time CPU Iterations
+ * ---------------------------------------------------------------
+ * BM_PREPROCESSING/1/0 59836 ns 59655 ns 11732
+ * BM_PREPROCESSING/1/1 66848 ns 66642 ns 10554
+ * BM_PREPROCESSING/1/2 20726 ns 20655 ns 33822
+ * BM_PREPROCESSING/1/3 5093 ns 5076 ns 137897
+ * BM_PREPROCESSING/2/0 117040 ns 116670 ns 5996
+ * BM_PREPROCESSING/2/1 120600 ns 120225 ns 5845
+ * BM_PREPROCESSING/2/2 38460 ns 38330 ns 18190
+ * BM_PREPROCESSING/2/3 6294 ns 6274 ns 111488
+ * BM_PREPROCESSING/3/0 232272 ns 231528 ns 3025
+ * BM_PREPROCESSING/3/1 226346 ns 225628 ns 3117
+ * BM_PREPROCESSING/3/2 75442 ns 75227 ns 9104
+ * BM_PREPROCESSING/3/3 9782 ns 9750 ns 71805
+ * BM_PREPROCESSING/4/0 290388 ns 289426 ns 2389
+ * BM_PREPROCESSING/4/1 279394 ns 278498 ns 2522
+ * BM_PREPROCESSING/4/2 94029 ns 93759 ns 7307
+ * BM_PREPROCESSING/4/3 11487 ns 11450 ns 61129
+ * BM_PREPROCESSING/5/0 347736 ns 346580 ns 2020
+ * BM_PREPROCESSING/5/1 331853 ns 330788 ns 2122
+ * BM_PREPROCESSING/5/2 112594 ns 112268 ns 6105
+ * BM_PREPROCESSING/5/3 13254 ns 13212 ns 52972
+ *******************************************************************/
+
+#include <audio_effects/effect_aec.h>
+#include <audio_effects/effect_agc.h>
+#include <array>
+#include <climits>
+#include <cstdlib>
+#include <random>
+#include <vector>
+#ifndef WEBRTC_LEGACY
+#include <audio_effects/effect_agc2.h>
+#endif
+#include <audio_effects/effect_ns.h>
+#include <benchmark/benchmark.h>
+#include <hardware/audio_effect.h>
+#include <log/log.h>
+#include <sys/stat.h>
+#include <system/audio.h>
+
+extern audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM;
+
+constexpr int kSampleRate = 16000;
+constexpr float kTenMilliSecVal = 0.01;
+constexpr unsigned int kStreamDelayMs = 0;
+constexpr effect_uuid_t kEffectUuids[] = {
+ // agc uuid
+ {0xaa8130e0, 0x66fc, 0x11e0, 0xbad0, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}},
+ // aec uuid
+ {0xbb392ec0, 0x8d4d, 0x11e0, 0xa896, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}},
+ // ns uuid
+ {0xc06c8400, 0x8e06, 0x11e0, 0x9cb6, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}},
+#ifndef WEBRTC_LEGACY
+ // agc2 uuid
+ {0x89f38e65, 0xd4d2, 0x4d64, 0xad0e, {0x2b, 0x3e, 0x79, 0x9e, 0xa8, 0x86}},
+#endif
+};
+constexpr size_t kNumEffectUuids = std::size(kEffectUuids);
+constexpr audio_channel_mask_t kChMasks[] = {
+ AUDIO_CHANNEL_IN_MONO, AUDIO_CHANNEL_IN_STEREO, AUDIO_CHANNEL_IN_2POINT0POINT2,
+ AUDIO_CHANNEL_IN_2POINT1POINT2, AUDIO_CHANNEL_IN_6,
+};
+constexpr size_t kNumChMasks = std::size(kChMasks);
+
+// types of pre processing modules
+enum PreProcId {
+ PREPROC_AGC, // Automatic Gain Control
+ PREPROC_AEC, // Acoustic Echo Canceler
+ PREPROC_NS, // Noise Suppressor
+#ifndef WEBRTC_LEGACY
+ PREPROC_AGC2, // Automatic Gain Control 2
+#endif
+ PREPROC_NUM_EFFECTS
+};
+
+int preProcCreateEffect(effect_handle_t* pEffectHandle, uint32_t effectType,
+ effect_config_t* pConfig, int sessionId, int ioId) {
+ if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.create_effect(&kEffectUuids[effectType],
+ sessionId, ioId, pEffectHandle);
+ status != 0) {
+ ALOGE("Audio Preprocessing create returned an error = %d\n", status);
+ return EXIT_FAILURE;
+ }
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ if (effectType == PREPROC_AEC) {
+ if (int status = (**pEffectHandle)
+ ->command(*pEffectHandle, EFFECT_CMD_SET_CONFIG_REVERSE,
+ sizeof(effect_config_t), pConfig, &replySize, &reply);
+ status != 0) {
+ ALOGE("Set config reverse command returned an error = %d\n", status);
+ return EXIT_FAILURE;
+ }
+ }
+ if (int status = (**pEffectHandle)
+ ->command(*pEffectHandle, EFFECT_CMD_SET_CONFIG,
+ sizeof(effect_config_t), pConfig, &replySize, &reply);
+ status != 0) {
+ ALOGE("Set config command returned an error = %d\n", status);
+ return EXIT_FAILURE;
+ }
+ return reply;
+}
+
+int preProcSetConfigParam(effect_handle_t effectHandle, uint32_t paramType, uint32_t paramValue) {
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ uint32_t paramData[2] = {paramType, paramValue};
+ effect_param_t* effectParam = (effect_param_t*)malloc(sizeof(*effectParam) + sizeof(paramData));
+ memcpy(&effectParam->data[0], ¶mData[0], sizeof(paramData));
+ effectParam->psize = sizeof(paramData[0]);
+ (*effectHandle)
+ ->command(effectHandle, EFFECT_CMD_SET_PARAM, sizeof(effect_param_t), effectParam,
+ &replySize, &reply);
+ free(effectParam);
+ return reply;
+}
+
+short preProcGetShortVal(float paramValue) {
+ return static_cast<short>(paramValue * std::numeric_limits<short>::max());
+}
+
+static void BM_PREPROCESSING(benchmark::State& state) {
+ const size_t chMask = kChMasks[state.range(0) - 1];
+ const size_t channelCount = audio_channel_count_from_in_mask(chMask);
+
+ PreProcId effectType = (PreProcId)state.range(1);
+
+ int32_t sessionId = 1;
+ int32_t ioId = 1;
+ effect_handle_t effectHandle = nullptr;
+ effect_config_t config{};
+ config.inputCfg.samplingRate = config.outputCfg.samplingRate = kSampleRate;
+ config.inputCfg.channels = config.outputCfg.channels = chMask;
+ config.inputCfg.format = config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+
+ if (int status = preProcCreateEffect(&effectHandle, state.range(1), &config, sessionId, ioId);
+ status != 0) {
+ ALOGE("Create effect call returned error %i", status);
+ return;
+ }
+
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ if (int status =
+ (*effectHandle)
+ ->command(effectHandle, EFFECT_CMD_ENABLE, 0, nullptr, &replySize, &reply);
+ status != 0) {
+ ALOGE("Command enable call returned error %d\n", reply);
+ return;
+ }
+
+ // Initialize input buffer with deterministic pseudo-random values
+ const int frameLength = (int)(kSampleRate * kTenMilliSecVal);
+ std::minstd_rand gen(chMask);
+ std::uniform_real_distribution<> dis(-1.0f, 1.0f);
+ std::vector<short> in(frameLength * channelCount);
+ for (auto& i : in) {
+ i = preProcGetShortVal(dis(gen));
+ }
+ std::vector<short> farIn(frameLength * channelCount);
+ for (auto& i : farIn) {
+ i = preProcGetShortVal(dis(gen));
+ }
+ std::vector<short> out(frameLength * channelCount);
+
+ // Run the test
+ for (auto _ : state) {
+ benchmark::DoNotOptimize(in.data());
+ benchmark::DoNotOptimize(out.data());
+ benchmark::DoNotOptimize(farIn.data());
+
+ audio_buffer_t inBuffer = {.frameCount = (size_t)frameLength, .s16 = in.data()};
+ audio_buffer_t outBuffer = {.frameCount = (size_t)frameLength, .s16 = out.data()};
+ audio_buffer_t farInBuffer = {.frameCount = (size_t)frameLength, .s16 = farIn.data()};
+
+ if (PREPROC_AEC == effectType) {
+ if (int status =
+ preProcSetConfigParam(effectHandle, AEC_PARAM_ECHO_DELAY, kStreamDelayMs);
+ status != 0) {
+ ALOGE("preProcSetConfigParam returned Error %d\n", status);
+ return;
+ }
+ }
+ if (int status = (*effectHandle)->process(effectHandle, &inBuffer, &outBuffer);
+ status != 0) {
+ ALOGE("\nError: Process i = %d returned with error %d\n", (int)state.range(1), status);
+ return;
+ }
+ if (PREPROC_AEC == effectType) {
+ if (int status =
+ (*effectHandle)->process_reverse(effectHandle, &farInBuffer, &outBuffer);
+ status != 0) {
+ ALOGE("\nError: Process reverse i = %d returned with error %d\n",
+ (int)state.range(1), status);
+ return;
+ }
+ }
+ }
+ benchmark::ClobberMemory();
+
+ state.SetComplexityN(state.range(0));
+
+ if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(effectHandle); status != 0) {
+ ALOGE("release_effect returned an error = %d\n", status);
+ return;
+ }
+}
+
+static void preprocessingArgs(benchmark::internal::Benchmark* b) {
+ for (int i = 1; i <= (int)kNumChMasks; i++) {
+ for (int j = 0; j < (int)kNumEffectUuids; ++j) {
+ b->Args({i, j});
+ }
+ }
+}
+
+BENCHMARK(BM_PREPROCESSING)->Apply(preprocessingArgs);
+
+BENCHMARK_MAIN();
diff --git a/media/libeffects/preprocessing/tests/PreProcessingTest.cpp b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
index 3244c1f..65b9469 100644
--- a/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
+++ b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
@@ -37,485 +37,465 @@
// types of pre processing modules
enum PreProcId {
- PREPROC_AGC, // Automatic Gain Control
+ PREPROC_AGC, // Automatic Gain Control
#ifndef WEBRTC_LEGACY
- PREPROC_AGC2, // Automatic Gain Control 2
+ PREPROC_AGC2, // Automatic Gain Control 2
#endif
- PREPROC_AEC, // Acoustic Echo Canceler
- PREPROC_NS, // Noise Suppressor
- PREPROC_NUM_EFFECTS
+ PREPROC_AEC, // Acoustic Echo Canceler
+ PREPROC_NS, // Noise Suppressor
+ PREPROC_NUM_EFFECTS
};
enum PreProcParams {
- ARG_HELP = 1,
- ARG_INPUT,
- ARG_OUTPUT,
- ARG_FAR,
- ARG_FS,
- ARG_CH_MASK,
- ARG_AGC_TGT_LVL,
- ARG_AGC_COMP_LVL,
- ARG_AEC_DELAY,
- ARG_NS_LVL,
+ ARG_HELP = 1,
+ ARG_INPUT,
+ ARG_OUTPUT,
+ ARG_FAR,
+ ARG_FS,
+ ARG_CH_MASK,
+ ARG_AGC_TGT_LVL,
+ ARG_AGC_COMP_LVL,
+ ARG_AEC_DELAY,
+ ARG_NS_LVL,
#ifndef WEBRTC_LEGACY
- ARG_AEC_MOBILE,
- ARG_AGC2_GAIN,
- ARG_AGC2_LVL,
- ARG_AGC2_SAT_MGN
+ ARG_AGC2_GAIN,
+ ARG_AGC2_LVL,
+ ARG_AGC2_SAT_MGN
#endif
};
struct preProcConfigParams_t {
- int samplingFreq = 16000;
- audio_channel_mask_t chMask = AUDIO_CHANNEL_IN_MONO;
- int nsLevel = 0; // a value between 0-3
- int agcTargetLevel = 3; // in dB
- int agcCompLevel = 9; // in dB
+ int samplingFreq = 16000;
+ audio_channel_mask_t chMask = AUDIO_CHANNEL_IN_MONO;
+ int nsLevel = 0; // a value between 0-3
+ int agcTargetLevel = 3; // in dB
+ int agcCompLevel = 9; // in dB
#ifndef WEBRTC_LEGACY
- float agc2Gain = 0.f; // in dB
- float agc2SaturationMargin = 2.f; // in dB
- int agc2Level = 0; // either kRms(0) or kPeak(1)
+ float agc2Gain = 0.f; // in dB
+ float agc2SaturationMargin = 2.f; // in dB
+ int agc2Level = 0; // either kRms(0) or kPeak(1)
#endif
- int aecDelay = 0; // in ms
+ int aecDelay = 0; // in ms
};
const effect_uuid_t kPreProcUuids[PREPROC_NUM_EFFECTS] = {
- {0xaa8130e0, 0x66fc, 0x11e0, 0xbad0, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // agc uuid
+ {0xaa8130e0, 0x66fc, 0x11e0, 0xbad0, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // agc uuid
#ifndef WEBRTC_LEGACY
- {0x89f38e65, 0xd4d2, 0x4d64, 0xad0e, {0x2b, 0x3e, 0x79, 0x9e, 0xa8, 0x86}}, // agc2 uuid
+ {0x89f38e65, 0xd4d2, 0x4d64, 0xad0e, {0x2b, 0x3e, 0x79, 0x9e, 0xa8, 0x86}}, // agc2 uuid
#endif
- {0xbb392ec0, 0x8d4d, 0x11e0, 0xa896, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // aec uuid
- {0xc06c8400, 0x8e06, 0x11e0, 0x9cb6, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // ns uuid
+ {0xbb392ec0, 0x8d4d, 0x11e0, 0xa896, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // aec uuid
+ {0xc06c8400, 0x8e06, 0x11e0, 0x9cb6, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // ns uuid
};
constexpr audio_channel_mask_t kPreProcConfigChMask[] = {
- AUDIO_CHANNEL_IN_MONO,
- AUDIO_CHANNEL_IN_STEREO,
- AUDIO_CHANNEL_IN_FRONT_BACK,
- AUDIO_CHANNEL_IN_6,
- AUDIO_CHANNEL_IN_2POINT0POINT2,
- AUDIO_CHANNEL_IN_2POINT1POINT2,
- AUDIO_CHANNEL_IN_3POINT0POINT2,
- AUDIO_CHANNEL_IN_3POINT1POINT2,
- AUDIO_CHANNEL_IN_5POINT1,
- AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO,
- AUDIO_CHANNEL_IN_VOICE_DNLINK_MONO,
- AUDIO_CHANNEL_IN_VOICE_CALL_MONO,
+ AUDIO_CHANNEL_IN_MONO,
+ AUDIO_CHANNEL_IN_STEREO,
+ AUDIO_CHANNEL_IN_FRONT_BACK,
+ AUDIO_CHANNEL_IN_6,
+ AUDIO_CHANNEL_IN_2POINT0POINT2,
+ AUDIO_CHANNEL_IN_2POINT1POINT2,
+ AUDIO_CHANNEL_IN_3POINT0POINT2,
+ AUDIO_CHANNEL_IN_3POINT1POINT2,
+ AUDIO_CHANNEL_IN_5POINT1,
+ AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO,
+ AUDIO_CHANNEL_IN_VOICE_DNLINK_MONO,
+ AUDIO_CHANNEL_IN_VOICE_CALL_MONO,
};
constexpr int kPreProcConfigChMaskCount = std::size(kPreProcConfigChMask);
void printUsage() {
- printf("\nUsage: ");
- printf("\n <executable> [options]\n");
- printf("\nwhere options are, ");
- printf("\n --input <inputfile>");
- printf("\n path to the input file");
- printf("\n --output <outputfile>");
- printf("\n path to the output file");
- printf("\n --help");
- printf("\n Prints this usage information");
- printf("\n --fs <sampling_freq>");
- printf("\n Sampling frequency in Hz, default 16000.");
- printf("\n -ch_mask <channel_mask>\n");
- printf("\n 0 - AUDIO_CHANNEL_IN_MONO");
- printf("\n 1 - AUDIO_CHANNEL_IN_STEREO");
- printf("\n 2 - AUDIO_CHANNEL_IN_FRONT_BACK");
- printf("\n 3 - AUDIO_CHANNEL_IN_6");
- printf("\n 4 - AUDIO_CHANNEL_IN_2POINT0POINT2");
- printf("\n 5 - AUDIO_CHANNEL_IN_2POINT1POINT2");
- printf("\n 6 - AUDIO_CHANNEL_IN_3POINT0POINT2");
- printf("\n 7 - AUDIO_CHANNEL_IN_3POINT1POINT2");
- printf("\n 8 - AUDIO_CHANNEL_IN_5POINT1");
- printf("\n 9 - AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO");
- printf("\n 10 - AUDIO_CHANNEL_IN_VOICE_DNLINK_MONO ");
- printf("\n 11 - AUDIO_CHANNEL_IN_VOICE_CALL_MONO ");
- printf("\n default 0");
- printf("\n --far <farend_file>");
- printf("\n Path to far-end file needed for echo cancellation");
- printf("\n --aec");
- printf("\n Enable Echo Cancellation, default disabled");
- printf("\n --ns");
- printf("\n Enable Noise Suppression, default disabled");
- printf("\n --agc");
- printf("\n Enable Gain Control, default disabled");
+ printf("\nUsage: ");
+ printf("\n <executable> [options]\n");
+ printf("\nwhere options are, ");
+ printf("\n --input <inputfile>");
+ printf("\n path to the input file");
+ printf("\n --output <outputfile>");
+ printf("\n path to the output file");
+ printf("\n --help");
+ printf("\n Prints this usage information");
+ printf("\n --fs <sampling_freq>");
+ printf("\n Sampling frequency in Hz, default 16000.");
+ printf("\n -ch_mask <channel_mask>\n");
+ printf("\n 0 - AUDIO_CHANNEL_IN_MONO");
+ printf("\n 1 - AUDIO_CHANNEL_IN_STEREO");
+ printf("\n 2 - AUDIO_CHANNEL_IN_FRONT_BACK");
+ printf("\n 3 - AUDIO_CHANNEL_IN_6");
+ printf("\n 4 - AUDIO_CHANNEL_IN_2POINT0POINT2");
+ printf("\n 5 - AUDIO_CHANNEL_IN_2POINT1POINT2");
+ printf("\n 6 - AUDIO_CHANNEL_IN_3POINT0POINT2");
+ printf("\n 7 - AUDIO_CHANNEL_IN_3POINT1POINT2");
+ printf("\n 8 - AUDIO_CHANNEL_IN_5POINT1");
+ printf("\n 9 - AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO");
+ printf("\n 10 - AUDIO_CHANNEL_IN_VOICE_DNLINK_MONO ");
+ printf("\n 11 - AUDIO_CHANNEL_IN_VOICE_CALL_MONO ");
+ printf("\n default 0");
+ printf("\n --far <farend_file>");
+ printf("\n Path to far-end file needed for echo cancellation");
+ printf("\n --aec");
+ printf("\n Enable Echo Cancellation, default disabled");
+ printf("\n --ns");
+ printf("\n Enable Noise Suppression, default disabled");
+ printf("\n --agc");
+ printf("\n Enable Gain Control, default disabled");
#ifndef WEBRTC_LEGACY
- printf("\n --agc2");
- printf("\n Enable Gain Controller 2, default disabled");
+ printf("\n --agc2");
+ printf("\n Enable Gain Controller 2, default disabled");
#endif
- printf("\n --ns_lvl <ns_level>");
- printf("\n Noise Suppression level in dB, default value 0dB");
- printf("\n --agc_tgt_lvl <target_level>");
- printf("\n AGC Target Level in dB, default value 3dB");
- printf("\n --agc_comp_lvl <comp_level>");
- printf("\n AGC Comp Level in dB, default value 9dB");
+ printf("\n --ns_lvl <ns_level>");
+ printf("\n Noise Suppression level in dB, default value 0dB");
+ printf("\n --agc_tgt_lvl <target_level>");
+ printf("\n AGC Target Level in dB, default value 3dB");
+ printf("\n --agc_comp_lvl <comp_level>");
+ printf("\n AGC Comp Level in dB, default value 9dB");
#ifndef WEBRTC_LEGACY
- printf("\n --agc2_gain <fixed_digital_gain>");
- printf("\n AGC Fixed Digital Gain in dB, default value 0dB");
- printf("\n --agc2_lvl <level_estimator>");
- printf("\n AGC Adaptive Digital Level Estimator, default value kRms");
- printf("\n --agc2_sat_mgn <saturation_margin>");
- printf("\n AGC Adaptive Digital Saturation Margin in dB, default value 2dB");
+ printf("\n --agc2_gain <fixed_digital_gain>");
+ printf("\n AGC Fixed Digital Gain in dB, default value 0dB");
+ printf("\n --agc2_lvl <level_estimator>");
+ printf("\n AGC Adaptive Digital Level Estimator, default value kRms");
+ printf("\n --agc2_sat_mgn <saturation_margin>");
+ printf("\n AGC Adaptive Digital Saturation Margin in dB, default value 2dB");
#endif
- printf("\n --aec_delay <delay>");
- printf("\n AEC delay value in ms, default value 0ms");
-#ifndef WEBRTC_LEGACY
- printf("\n --aec_mobile");
- printf("\n Enable mobile mode of echo canceller, default disabled");
-#endif
- printf("\n");
+ printf("\n --aec_delay <delay>");
+ printf("\n AEC delay value in ms, default value 0ms");
+ printf("\n");
}
constexpr float kTenMilliSecVal = 0.01;
-int preProcCreateEffect(effect_handle_t *pEffectHandle, uint32_t effectType,
- effect_config_t *pConfig, int sessionId, int ioId) {
- if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.create_effect(&kPreProcUuids[effectType],
- sessionId, ioId, pEffectHandle);
- status != 0) {
- ALOGE("Audio Preprocessing create returned an error = %d\n", status);
- return EXIT_FAILURE;
- }
- int reply = 0;
- uint32_t replySize = sizeof(reply);
- if (effectType == PREPROC_AEC) {
+int preProcCreateEffect(effect_handle_t* pEffectHandle, uint32_t effectType,
+ effect_config_t* pConfig, int sessionId, int ioId) {
+ if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.create_effect(&kPreProcUuids[effectType],
+ sessionId, ioId, pEffectHandle);
+ status != 0) {
+ ALOGE("Audio Preprocessing create returned an error = %d\n", status);
+ return EXIT_FAILURE;
+ }
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ if (effectType == PREPROC_AEC) {
+ (**pEffectHandle)
+ ->command(*pEffectHandle, EFFECT_CMD_SET_CONFIG_REVERSE, sizeof(effect_config_t),
+ pConfig, &replySize, &reply);
+ }
(**pEffectHandle)
- ->command(*pEffectHandle, EFFECT_CMD_SET_CONFIG_REVERSE, sizeof(effect_config_t), pConfig,
- &replySize, &reply);
- }
- (**pEffectHandle)
- ->command(*pEffectHandle, EFFECT_CMD_SET_CONFIG, sizeof(effect_config_t), pConfig,
- &replySize, &reply);
- return reply;
+ ->command(*pEffectHandle, EFFECT_CMD_SET_CONFIG, sizeof(effect_config_t), pConfig,
+ &replySize, &reply);
+ return reply;
}
int preProcSetConfigParam(uint32_t paramType, uint32_t paramValue, effect_handle_t effectHandle) {
- int reply = 0;
- uint32_t replySize = sizeof(reply);
- uint32_t paramData[2] = {paramType, paramValue};
- effect_param_t *effectParam =
- (effect_param_t *)malloc(sizeof(*effectParam) + sizeof(paramData));
- memcpy(&effectParam->data[0], ¶mData[0], sizeof(paramData));
- effectParam->psize = sizeof(paramData[0]);
- (*effectHandle)
- ->command(effectHandle, EFFECT_CMD_SET_PARAM, sizeof(effect_param_t), effectParam,
- &replySize, &reply);
- free(effectParam);
- return reply;
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ uint32_t paramData[2] = {paramType, paramValue};
+ effect_param_t* effectParam = (effect_param_t*)malloc(sizeof(*effectParam) + sizeof(paramData));
+ memcpy(&effectParam->data[0], ¶mData[0], sizeof(paramData));
+ effectParam->psize = sizeof(paramData[0]);
+ (*effectHandle)
+ ->command(effectHandle, EFFECT_CMD_SET_PARAM, sizeof(effect_param_t), effectParam,
+ &replySize, &reply);
+ free(effectParam);
+ return reply;
}
-int main(int argc, const char *argv[]) {
- if (argc == 1) {
- printUsage();
- return EXIT_FAILURE;
- }
- const char *inputFile = nullptr;
- const char *outputFile = nullptr;
- const char *farFile = nullptr;
- int effectEn[PREPROC_NUM_EFFECTS] = {0};
-#ifndef WEBRTC_LEGACY
- int aecMobileMode = 0;
-#endif
-
- const option long_opts[] = {
- {"help", no_argument, nullptr, ARG_HELP},
- {"input", required_argument, nullptr, ARG_INPUT},
- {"output", required_argument, nullptr, ARG_OUTPUT},
- {"far", required_argument, nullptr, ARG_FAR},
- {"fs", required_argument, nullptr, ARG_FS},
- {"ch_mask", required_argument, nullptr, ARG_CH_MASK},
- {"agc_tgt_lvl", required_argument, nullptr, ARG_AGC_TGT_LVL},
- {"agc_comp_lvl", required_argument, nullptr, ARG_AGC_COMP_LVL},
-#ifndef WEBRTC_LEGACY
- {"agc2_gain", required_argument, nullptr, ARG_AGC2_GAIN},
- {"agc2_lvl", required_argument, nullptr, ARG_AGC2_LVL},
- {"agc2_sat_mgn", required_argument, nullptr, ARG_AGC2_SAT_MGN},
-#endif
- {"aec_delay", required_argument, nullptr, ARG_AEC_DELAY},
- {"ns_lvl", required_argument, nullptr, ARG_NS_LVL},
- {"aec", no_argument, &effectEn[PREPROC_AEC], 1},
- {"agc", no_argument, &effectEn[PREPROC_AGC], 1},
-#ifndef WEBRTC_LEGACY
- {"agc2", no_argument, &effectEn[PREPROC_AGC2], 1},
-#endif
- {"ns", no_argument, &effectEn[PREPROC_NS], 1},
-#ifndef WEBRTC_LEGACY
- {"aec_mobile", no_argument, &aecMobileMode, 1},
-#endif
- {nullptr, 0, nullptr, 0},
- };
- struct preProcConfigParams_t preProcCfgParams {};
-
- while (true) {
- const int opt = getopt_long(argc, (char *const *)argv, "i:o:", long_opts, nullptr);
- if (opt == -1) {
- break;
- }
- switch (opt) {
- case ARG_HELP:
+int main(int argc, const char* argv[]) {
+ if (argc == 1) {
printUsage();
- return 0;
- case ARG_INPUT: {
- inputFile = (char *)optarg;
- break;
- }
- case ARG_OUTPUT: {
- outputFile = (char *)optarg;
- break;
- }
- case ARG_FAR: {
- farFile = (char *)optarg;
- break;
- }
- case ARG_FS: {
- preProcCfgParams.samplingFreq = atoi(optarg);
- break;
- }
- case ARG_CH_MASK: {
- int chMaskIdx = atoi(optarg);
- if (chMaskIdx < 0 or chMaskIdx > kPreProcConfigChMaskCount) {
- ALOGE("Channel Mask index not in correct range\n");
- printUsage();
- return EXIT_FAILURE;
- }
- preProcCfgParams.chMask = kPreProcConfigChMask[chMaskIdx];
- break;
- }
- case ARG_AGC_TGT_LVL: {
- preProcCfgParams.agcTargetLevel = atoi(optarg);
- break;
- }
- case ARG_AGC_COMP_LVL: {
- preProcCfgParams.agcCompLevel = atoi(optarg);
- break;
- }
-#ifndef WEBRTC_LEGACY
- case ARG_AGC2_GAIN: {
- preProcCfgParams.agc2Gain = atof(optarg);
- break;
- }
- case ARG_AGC2_LVL: {
- preProcCfgParams.agc2Level = atoi(optarg);
- break;
- }
- case ARG_AGC2_SAT_MGN: {
- preProcCfgParams.agc2SaturationMargin = atof(optarg);
- break;
- }
-#endif
- case ARG_AEC_DELAY: {
- preProcCfgParams.aecDelay = atoi(optarg);
- break;
- }
- case ARG_NS_LVL: {
- preProcCfgParams.nsLevel = atoi(optarg);
- break;
- }
- default:
- break;
- }
- }
-
- if (inputFile == nullptr) {
- ALOGE("Error: missing input file\n");
- printUsage();
- return EXIT_FAILURE;
- }
-
- std::unique_ptr<FILE, decltype(&fclose)> inputFp(fopen(inputFile, "rb"), &fclose);
- if (inputFp == nullptr) {
- ALOGE("Cannot open input file %s\n", inputFile);
- return EXIT_FAILURE;
- }
-
- std::unique_ptr<FILE, decltype(&fclose)> farFp(fopen(farFile, "rb"), &fclose);
- std::unique_ptr<FILE, decltype(&fclose)> outputFp(fopen(outputFile, "wb"), &fclose);
- if (effectEn[PREPROC_AEC]) {
- if (farFile == nullptr) {
- ALOGE("Far end signal file required for echo cancellation \n");
- return EXIT_FAILURE;
- }
- if (farFp == nullptr) {
- ALOGE("Cannot open far end stream file %s\n", farFile);
- return EXIT_FAILURE;
- }
- struct stat statInput, statFar;
- (void)fstat(fileno(inputFp.get()), &statInput);
- (void)fstat(fileno(farFp.get()), &statFar);
- if (statInput.st_size != statFar.st_size) {
- ALOGE("Near and far end signals are of different sizes");
- return EXIT_FAILURE;
- }
- }
- if (outputFile != nullptr && outputFp == nullptr) {
- ALOGE("Cannot open output file %s\n", outputFile);
- return EXIT_FAILURE;
- }
-
- int32_t sessionId = 1;
- int32_t ioId = 1;
- effect_handle_t effectHandle[PREPROC_NUM_EFFECTS] = {nullptr};
- effect_config_t config;
- config.inputCfg.samplingRate = config.outputCfg.samplingRate = preProcCfgParams.samplingFreq;
- config.inputCfg.channels = config.outputCfg.channels = preProcCfgParams.chMask;
- config.inputCfg.format = config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
-
- // Create all the effect handles
- for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
- if (int status = preProcCreateEffect(&effectHandle[i], i, &config, sessionId, ioId);
- status != 0) {
- ALOGE("Create effect call returned error %i", status);
- return EXIT_FAILURE;
- }
- }
-
- for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
- if (effectEn[i] == 1) {
- int reply = 0;
- uint32_t replySize = sizeof(reply);
- (*effectHandle[i])
- ->command(effectHandle[i], EFFECT_CMD_ENABLE, 0, nullptr, &replySize, &reply);
- if (reply != 0) {
- ALOGE("Command enable call returned error %d\n", reply);
return EXIT_FAILURE;
- }
}
- }
+ const char* inputFile = nullptr;
+ const char* outputFile = nullptr;
+ const char* farFile = nullptr;
+ int effectEn[PREPROC_NUM_EFFECTS] = {0};
- // Set Config Params of the effects
- if (effectEn[PREPROC_AGC]) {
- if (int status = preProcSetConfigParam(AGC_PARAM_TARGET_LEVEL,
- (uint32_t)preProcCfgParams.agcTargetLevel,
- effectHandle[PREPROC_AGC]);
- status != 0) {
- ALOGE("Invalid AGC Target Level. Error %d\n", status);
- return EXIT_FAILURE;
- }
- if (int status =
- preProcSetConfigParam(AGC_PARAM_COMP_GAIN, (uint32_t)preProcCfgParams.agcCompLevel,
- effectHandle[PREPROC_AGC]);
- status != 0) {
- ALOGE("Invalid AGC Comp Gain. Error %d\n", status);
- return EXIT_FAILURE;
- }
- }
+ const option long_opts[] = {
+ {"help", no_argument, nullptr, ARG_HELP},
+ {"input", required_argument, nullptr, ARG_INPUT},
+ {"output", required_argument, nullptr, ARG_OUTPUT},
+ {"far", required_argument, nullptr, ARG_FAR},
+ {"fs", required_argument, nullptr, ARG_FS},
+ {"ch_mask", required_argument, nullptr, ARG_CH_MASK},
+ {"agc_tgt_lvl", required_argument, nullptr, ARG_AGC_TGT_LVL},
+ {"agc_comp_lvl", required_argument, nullptr, ARG_AGC_COMP_LVL},
#ifndef WEBRTC_LEGACY
- if (effectEn[PREPROC_AGC2]) {
- if (int status = preProcSetConfigParam(AGC2_PARAM_FIXED_DIGITAL_GAIN,
- (float)preProcCfgParams.agc2Gain,
- effectHandle[PREPROC_AGC2]);
- status != 0) {
- ALOGE("Invalid AGC2 Fixed Digital Gain. Error %d\n", status);
- return EXIT_FAILURE;
- }
- if (int status = preProcSetConfigParam(AGC2_PARAM_ADAPT_DIGI_LEVEL_ESTIMATOR,
- (uint32_t)preProcCfgParams.agc2Level,
- effectHandle[PREPROC_AGC2]);
- status != 0) {
- ALOGE("Invalid AGC2 Level Estimator. Error %d\n", status);
- return EXIT_FAILURE;
- }
- if (int status = preProcSetConfigParam(AGC2_PARAM_ADAPT_DIGI_EXTRA_SATURATION_MARGIN,
- (float)preProcCfgParams.agc2SaturationMargin,
- effectHandle[PREPROC_AGC2]);
- status != 0) {
- ALOGE("Invalid AGC2 Saturation Margin. Error %d\n", status);
- return EXIT_FAILURE;
- }
- }
+ {"agc2_gain", required_argument, nullptr, ARG_AGC2_GAIN},
+ {"agc2_lvl", required_argument, nullptr, ARG_AGC2_LVL},
+ {"agc2_sat_mgn", required_argument, nullptr, ARG_AGC2_SAT_MGN},
#endif
- if (effectEn[PREPROC_NS]) {
- if (int status = preProcSetConfigParam(NS_PARAM_LEVEL, (uint32_t)preProcCfgParams.nsLevel,
- effectHandle[PREPROC_NS]);
- status != 0) {
- ALOGE("Invalid Noise Suppression level Error %d\n", status);
- return EXIT_FAILURE;
- }
- }
+ {"aec_delay", required_argument, nullptr, ARG_AEC_DELAY},
+ {"ns_lvl", required_argument, nullptr, ARG_NS_LVL},
+ {"aec", no_argument, &effectEn[PREPROC_AEC], 1},
+ {"agc", no_argument, &effectEn[PREPROC_AGC], 1},
#ifndef WEBRTC_LEGACY
- if (effectEn[PREPROC_AEC]) {
- if (int status = preProcSetConfigParam(AEC_PARAM_MOBILE_MODE, (uint32_t)aecMobileMode,
- effectHandle[PREPROC_AEC]);
- status != 0) {
- ALOGE("Invalid AEC mobile mode value %d\n", status);
- return EXIT_FAILURE;
- }
- }
+ {"agc2", no_argument, &effectEn[PREPROC_AGC2], 1},
#endif
+ {"ns", no_argument, &effectEn[PREPROC_NS], 1},
+ {nullptr, 0, nullptr, 0},
+ };
+ struct preProcConfigParams_t preProcCfgParams {};
- // Process Call
- const int frameLength = (int)(preProcCfgParams.samplingFreq * kTenMilliSecVal);
- const int ioChannelCount = audio_channel_count_from_in_mask(preProcCfgParams.chMask);
- const int ioFrameSize = ioChannelCount * sizeof(short);
- int frameCounter = 0;
- while (true) {
- std::vector<short> in(frameLength * ioChannelCount);
- std::vector<short> out(frameLength * ioChannelCount);
- std::vector<short> farIn(frameLength * ioChannelCount);
- size_t samplesRead = fread(in.data(), ioFrameSize, frameLength, inputFp.get());
- if (samplesRead == 0) {
- break;
+ while (true) {
+ const int opt = getopt_long(argc, (char* const*)argv, "i:o:", long_opts, nullptr);
+ if (opt == -1) {
+ break;
+ }
+ switch (opt) {
+ case ARG_HELP:
+ printUsage();
+ return 0;
+ case ARG_INPUT: {
+ inputFile = (char*)optarg;
+ break;
+ }
+ case ARG_OUTPUT: {
+ outputFile = (char*)optarg;
+ break;
+ }
+ case ARG_FAR: {
+ farFile = (char*)optarg;
+ break;
+ }
+ case ARG_FS: {
+ preProcCfgParams.samplingFreq = atoi(optarg);
+ break;
+ }
+ case ARG_CH_MASK: {
+ int chMaskIdx = atoi(optarg);
+ if (chMaskIdx < 0 or chMaskIdx > kPreProcConfigChMaskCount) {
+ ALOGE("Channel Mask index not in correct range\n");
+ printUsage();
+ return EXIT_FAILURE;
+ }
+ preProcCfgParams.chMask = kPreProcConfigChMask[chMaskIdx];
+ break;
+ }
+ case ARG_AGC_TGT_LVL: {
+ preProcCfgParams.agcTargetLevel = atoi(optarg);
+ break;
+ }
+ case ARG_AGC_COMP_LVL: {
+ preProcCfgParams.agcCompLevel = atoi(optarg);
+ break;
+ }
+#ifndef WEBRTC_LEGACY
+ case ARG_AGC2_GAIN: {
+ preProcCfgParams.agc2Gain = atof(optarg);
+ break;
+ }
+ case ARG_AGC2_LVL: {
+ preProcCfgParams.agc2Level = atoi(optarg);
+ break;
+ }
+ case ARG_AGC2_SAT_MGN: {
+ preProcCfgParams.agc2SaturationMargin = atof(optarg);
+ break;
+ }
+#endif
+ case ARG_AEC_DELAY: {
+ preProcCfgParams.aecDelay = atoi(optarg);
+ break;
+ }
+ case ARG_NS_LVL: {
+ preProcCfgParams.nsLevel = atoi(optarg);
+ break;
+ }
+ default:
+ break;
+ }
}
- audio_buffer_t inputBuffer, outputBuffer;
- audio_buffer_t farInBuffer{};
- inputBuffer.frameCount = samplesRead;
- outputBuffer.frameCount = samplesRead;
- inputBuffer.s16 = in.data();
- outputBuffer.s16 = out.data();
- if (farFp != nullptr) {
- samplesRead = fread(farIn.data(), ioFrameSize, frameLength, farFp.get());
- if (samplesRead == 0) {
- break;
- }
- farInBuffer.frameCount = samplesRead;
- farInBuffer.s16 = farIn.data();
+ if (inputFile == nullptr) {
+ ALOGE("Error: missing input file\n");
+ printUsage();
+ return EXIT_FAILURE;
+ }
+
+ std::unique_ptr<FILE, decltype(&fclose)> inputFp(fopen(inputFile, "rb"), &fclose);
+ if (inputFp == nullptr) {
+ ALOGE("Cannot open input file %s\n", inputFile);
+ return EXIT_FAILURE;
+ }
+
+ std::unique_ptr<FILE, decltype(&fclose)> farFp(fopen(farFile, "rb"), &fclose);
+ std::unique_ptr<FILE, decltype(&fclose)> outputFp(fopen(outputFile, "wb"), &fclose);
+ if (effectEn[PREPROC_AEC]) {
+ if (farFile == nullptr) {
+ ALOGE("Far end signal file required for echo cancellation \n");
+ return EXIT_FAILURE;
+ }
+ if (farFp == nullptr) {
+ ALOGE("Cannot open far end stream file %s\n", farFile);
+ return EXIT_FAILURE;
+ }
+ struct stat statInput, statFar;
+ (void)fstat(fileno(inputFp.get()), &statInput);
+ (void)fstat(fileno(farFp.get()), &statFar);
+ if (statInput.st_size != statFar.st_size) {
+ ALOGE("Near and far end signals are of different sizes");
+ return EXIT_FAILURE;
+ }
+ }
+ if (outputFile != nullptr && outputFp == nullptr) {
+ ALOGE("Cannot open output file %s\n", outputFile);
+ return EXIT_FAILURE;
+ }
+
+ int32_t sessionId = 1;
+ int32_t ioId = 1;
+ effect_handle_t effectHandle[PREPROC_NUM_EFFECTS] = {nullptr};
+ effect_config_t config;
+ config.inputCfg.samplingRate = config.outputCfg.samplingRate = preProcCfgParams.samplingFreq;
+ config.inputCfg.channels = config.outputCfg.channels = preProcCfgParams.chMask;
+ config.inputCfg.format = config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+
+ // Create all the effect handles
+ for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
+ if (int status = preProcCreateEffect(&effectHandle[i], i, &config, sessionId, ioId);
+ status != 0) {
+ ALOGE("Create effect call returned error %i", status);
+ return EXIT_FAILURE;
+ }
}
for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
- if (effectEn[i] == 1) {
- if (i == PREPROC_AEC) {
- if (int status =
- preProcSetConfigParam(AEC_PARAM_ECHO_DELAY, (uint32_t)preProcCfgParams.aecDelay,
- effectHandle[PREPROC_AEC]);
- status != 0) {
- ALOGE("preProcSetConfigParam returned Error %d\n", status);
- return EXIT_FAILURE;
- }
+ if (effectEn[i] == 1) {
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ (*effectHandle[i])
+ ->command(effectHandle[i], EFFECT_CMD_ENABLE, 0, nullptr, &replySize, &reply);
+ if (reply != 0) {
+ ALOGE("Command enable call returned error %d\n", reply);
+ return EXIT_FAILURE;
+ }
}
- if (int status =
- (*effectHandle[i])->process(effectHandle[i], &inputBuffer, &outputBuffer);
+ }
+
+ // Set Config Params of the effects
+ if (effectEn[PREPROC_AGC]) {
+ if (int status = preProcSetConfigParam(AGC_PARAM_TARGET_LEVEL,
+ (uint32_t)preProcCfgParams.agcTargetLevel,
+ effectHandle[PREPROC_AGC]);
status != 0) {
- ALOGE("\nError: Process i = %d returned with error %d\n", i, status);
- return EXIT_FAILURE;
- }
- if (i == PREPROC_AEC) {
- if (int status = (*effectHandle[i])
- ->process_reverse(effectHandle[i], &farInBuffer, &outputBuffer);
- status != 0) {
- ALOGE("\nError: Process reverse i = %d returned with error %d\n", i, status);
+ ALOGE("Invalid AGC Target Level. Error %d\n", status);
return EXIT_FAILURE;
- }
}
- }
+ if (int status = preProcSetConfigParam(AGC_PARAM_COMP_GAIN,
+ (uint32_t)preProcCfgParams.agcCompLevel,
+ effectHandle[PREPROC_AGC]);
+ status != 0) {
+ ALOGE("Invalid AGC Comp Gain. Error %d\n", status);
+ return EXIT_FAILURE;
+ }
}
- if (outputFp != nullptr) {
- size_t samplesWritten =
- fwrite(out.data(), ioFrameSize, outputBuffer.frameCount, outputFp.get());
- if (samplesWritten != outputBuffer.frameCount) {
- ALOGE("\nError: Output file writing failed");
- break;
- }
+#ifndef WEBRTC_LEGACY
+ if (effectEn[PREPROC_AGC2]) {
+ if (int status = preProcSetConfigParam(AGC2_PARAM_FIXED_DIGITAL_GAIN,
+ (float)preProcCfgParams.agc2Gain,
+ effectHandle[PREPROC_AGC2]);
+ status != 0) {
+ ALOGE("Invalid AGC2 Fixed Digital Gain. Error %d\n", status);
+ return EXIT_FAILURE;
+ }
+ if (int status = preProcSetConfigParam(AGC2_PARAM_ADAPT_DIGI_LEVEL_ESTIMATOR,
+ (uint32_t)preProcCfgParams.agc2Level,
+ effectHandle[PREPROC_AGC2]);
+ status != 0) {
+ ALOGE("Invalid AGC2 Level Estimator. Error %d\n", status);
+ return EXIT_FAILURE;
+ }
+ if (int status = preProcSetConfigParam(AGC2_PARAM_ADAPT_DIGI_EXTRA_SATURATION_MARGIN,
+ (float)preProcCfgParams.agc2SaturationMargin,
+ effectHandle[PREPROC_AGC2]);
+ status != 0) {
+ ALOGE("Invalid AGC2 Saturation Margin. Error %d\n", status);
+ return EXIT_FAILURE;
+ }
}
- frameCounter += frameLength;
- }
- // Release all the effect handles created
- for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
- if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(effectHandle[i]);
- status != 0) {
- ALOGE("Audio Preprocessing release returned an error = %d\n", status);
- return EXIT_FAILURE;
+#endif
+ if (effectEn[PREPROC_NS]) {
+ if (int status = preProcSetConfigParam(NS_PARAM_LEVEL, (uint32_t)preProcCfgParams.nsLevel,
+ effectHandle[PREPROC_NS]);
+ status != 0) {
+ ALOGE("Invalid Noise Suppression level Error %d\n", status);
+ return EXIT_FAILURE;
+ }
}
- }
- return EXIT_SUCCESS;
+
+ // Process Call
+ const int frameLength = (int)(preProcCfgParams.samplingFreq * kTenMilliSecVal);
+ const int ioChannelCount = audio_channel_count_from_in_mask(preProcCfgParams.chMask);
+ const int ioFrameSize = ioChannelCount * sizeof(short);
+ int frameCounter = 0;
+ while (true) {
+ std::vector<short> in(frameLength * ioChannelCount);
+ std::vector<short> out(frameLength * ioChannelCount);
+ std::vector<short> farIn(frameLength * ioChannelCount);
+ size_t samplesRead = fread(in.data(), ioFrameSize, frameLength, inputFp.get());
+ if (samplesRead == 0) {
+ break;
+ }
+ audio_buffer_t inputBuffer, outputBuffer;
+ audio_buffer_t farInBuffer{};
+ inputBuffer.frameCount = samplesRead;
+ outputBuffer.frameCount = samplesRead;
+ inputBuffer.s16 = in.data();
+ outputBuffer.s16 = out.data();
+
+ if (farFp != nullptr) {
+ samplesRead = fread(farIn.data(), ioFrameSize, frameLength, farFp.get());
+ if (samplesRead == 0) {
+ break;
+ }
+ farInBuffer.frameCount = samplesRead;
+ farInBuffer.s16 = farIn.data();
+ }
+
+ for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
+ if (effectEn[i] == 1) {
+ if (i == PREPROC_AEC) {
+ if (int status = preProcSetConfigParam(AEC_PARAM_ECHO_DELAY,
+ (uint32_t)preProcCfgParams.aecDelay,
+ effectHandle[PREPROC_AEC]);
+ status != 0) {
+ ALOGE("preProcSetConfigParam returned Error %d\n", status);
+ return EXIT_FAILURE;
+ }
+ }
+ if (int status = (*effectHandle[i])
+ ->process(effectHandle[i], &inputBuffer, &outputBuffer);
+ status != 0) {
+ ALOGE("\nError: Process i = %d returned with error %d\n", i, status);
+ return EXIT_FAILURE;
+ }
+ if (i == PREPROC_AEC) {
+ if (int status = (*effectHandle[i])
+ ->process_reverse(effectHandle[i], &farInBuffer,
+ &outputBuffer);
+ status != 0) {
+ ALOGE("\nError: Process reverse i = %d returned with error %d\n", i,
+ status);
+ return EXIT_FAILURE;
+ }
+ }
+ }
+ }
+ if (outputFp != nullptr) {
+ size_t samplesWritten =
+ fwrite(out.data(), ioFrameSize, outputBuffer.frameCount, outputFp.get());
+ if (samplesWritten != outputBuffer.frameCount) {
+ ALOGE("\nError: Output file writing failed");
+ break;
+ }
+ }
+ frameCounter += frameLength;
+ }
+ // Release all the effect handles created
+ for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
+ if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(effectHandle[i]);
+ status != 0) {
+ ALOGE("Audio Preprocessing release returned an error = %d\n", status);
+ return EXIT_FAILURE;
+ }
+ }
+ return EXIT_SUCCESS;
}
diff --git a/media/libmediahelper/TypeConverter.cpp b/media/libmediahelper/TypeConverter.cpp
index 705959a..876dc45 100644
--- a/media/libmediahelper/TypeConverter.cpp
+++ b/media/libmediahelper/TypeConverter.cpp
@@ -32,18 +32,21 @@
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
- MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
+ // TODO(mnaganov): Remove from here, use 'audio_is_bluetooth_out_sco_device' function.
+ { "AUDIO_DEVICE_OUT_ALL_SCO", static_cast<audio_devices_t>(AUDIO_DEVICE_OUT_ALL_SCO) },
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
- MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
+ // TODO(mnaganov): Remove from here, use 'audio_is_a2dp_out_device' function.
+ { "AUDIO_DEVICE_OUT_ALL_A2DP", static_cast<audio_devices_t>(AUDIO_DEVICE_OUT_ALL_A2DP) },
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_HDMI),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
- MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
+ // TODO(mnaganov): Remove from here, use 'audio_is_usb_out_device' function.
+ { "AUDIO_DEVICE_OUT_ALL_USB", static_cast<audio_devices_t>(AUDIO_DEVICE_OUT_ALL_USB) },
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_LINE),
@@ -72,7 +75,8 @@
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_AMBIENT),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
- MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
+ // TODO(mnaganov): Remove from here, use 'audio_is_bluetooth_in_sco_device' function.
+ { "AUDIO_DEVICE_IN_ALL_SCO", static_cast<audio_devices_t>(AUDIO_DEVICE_IN_ALL_SCO) },
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_HDMI),
@@ -85,7 +89,8 @@
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
- MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_ALL_USB),
+ // TODO(mnaganov): Remove from here, use 'audio_is_usb_in_device' function.
+ { "AUDIO_DEVICE_IN_ALL_USB", static_cast<audio_devices_t>(AUDIO_DEVICE_IN_ALL_USB) },
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_LINE),
diff --git a/media/libmediametrics/Android.bp b/media/libmediametrics/Android.bp
index 03068c7..ba84761 100644
--- a/media/libmediametrics/Android.bp
+++ b/media/libmediametrics/Android.bp
@@ -3,7 +3,7 @@
export_include_dirs: ["include"],
}
-cc_library_shared {
+cc_library {
name: "libmediametrics",
srcs: [
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 7897959..71beceb 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -125,6 +125,7 @@
ALOGV("Constructor");
+ mMetricsItem = NULL;
mAnalyticsDirty = false;
reset();
}
@@ -199,10 +200,12 @@
void StagefrightRecorder::flushAndResetMetrics(bool reinitialize) {
ALOGV("flushAndResetMetrics");
// flush anything we have, maybe setup a new record
- if (mAnalyticsDirty && mMetricsItem != NULL) {
- updateMetrics();
- if (mMetricsItem->count() > 0) {
- mMetricsItem->selfrecord();
+ if (mMetricsItem != NULL) {
+ if (mAnalyticsDirty) {
+ updateMetrics();
+ if (mMetricsItem->count() > 0) {
+ mMetricsItem->selfrecord();
+ }
}
delete mMetricsItem;
mMetricsItem = NULL;
@@ -1113,7 +1116,7 @@
if (mPrivacySensitive == PRIVACY_SENSITIVE_DEFAULT) {
if (attr.source == AUDIO_SOURCE_VOICE_COMMUNICATION
|| attr.source == AUDIO_SOURCE_CAMCORDER) {
- attr.flags |= AUDIO_FLAG_CAPTURE_PRIVATE;
+ attr.flags = static_cast<audio_flags_mask_t>(attr.flags | AUDIO_FLAG_CAPTURE_PRIVATE);
mPrivacySensitive = PRIVACY_SENSITIVE_ENABLED;
} else {
mPrivacySensitive = PRIVACY_SENSITIVE_DISABLED;
@@ -1129,7 +1132,7 @@
return NULL;
}
if (mPrivacySensitive == PRIVACY_SENSITIVE_ENABLED) {
- attr.flags |= AUDIO_FLAG_CAPTURE_PRIVATE;
+ attr.flags = static_cast<audio_flags_mask_t>(attr.flags | AUDIO_FLAG_CAPTURE_PRIVATE);
}
}
diff --git a/media/libmediaplayerservice/include/MediaPlayerInterface.h b/media/libmediaplayerservice/include/MediaPlayerInterface.h
index 436cb31..81da5b9 100644
--- a/media/libmediaplayerservice/include/MediaPlayerInterface.h
+++ b/media/libmediaplayerservice/include/MediaPlayerInterface.h
@@ -60,7 +60,7 @@
#define DEFAULT_AUDIOSINK_SAMPLERATE 44100
// when the channel mask isn't known, use the channel count to derive a mask in AudioSink::open()
-#define CHANNEL_MASK_USE_CHANNEL_ORDER 0
+#define CHANNEL_MASK_USE_CHANNEL_ORDER AUDIO_CHANNEL_NONE
// duration below which we do not allow deep audio buffering
#define AUDIO_SINK_MIN_DEEP_BUFFER_DURATION_US 5000000
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 7e8fe45..6a8c708 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -1933,11 +1933,12 @@
int32_t numChannels;
CHECK(format->findInt32("channel-count", &numChannels));
- int32_t channelMask;
- if (!format->findInt32("channel-mask", &channelMask)) {
- // signal to the AudioSink to derive the mask from count.
- channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
- }
+ 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;
int32_t sampleRate;
CHECK(format->findInt32("sample-rate", &sampleRate));
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index ffe3052..bc84d78 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -5320,6 +5320,34 @@
if (mChannelMaskPresent) {
notify->setInt32("channel-mask", mChannelMask);
}
+
+ if (!mIsEncoder && portIndex == kPortIndexOutput) {
+ AString mime;
+ if (mConfigFormat->findString("mime", &mime)
+ && !strcasecmp(MEDIA_MIMETYPE_AUDIO_AAC, mime.c_str())) {
+
+ OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE presentation;
+ InitOMXParams(&presentation);
+ err = mOMXNode->getParameter(
+ (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAacDrcPresentation,
+ &presentation, sizeof(presentation));
+ if (err != OK) {
+ return err;
+ }
+ notify->setInt32("aac-encoded-target-level",
+ presentation.nEncodedTargetLevel);
+ notify->setInt32("aac-drc-cut-level", presentation.nDrcCut);
+ notify->setInt32("aac-drc-boost-level", presentation.nDrcBoost);
+ notify->setInt32("aac-drc-heavy-compression",
+ presentation.nHeavyCompression);
+ notify->setInt32("aac-target-ref-level",
+ presentation.nTargetReferenceLevel);
+ notify->setInt32("aac-drc-effect-type", presentation.nDrcEffectType);
+ notify->setInt32("aac-drc-album-mode", presentation.nDrcAlbumMode);
+ notify->setInt32("aac-drc-output-loudness",
+ presentation.nDrcOutputLoudness);
+ }
+ }
break;
}
@@ -7767,6 +7795,58 @@
// Ignore errors as failure is expected for codecs that aren't video encoders.
(void)configureTemporalLayers(params, false /* inConfigure */, mOutputFormat);
+ AString mime;
+ if (!mIsEncoder
+ && (mConfigFormat->findString("mime", &mime))
+ && !strcasecmp(MEDIA_MIMETYPE_AUDIO_AAC, mime.c_str())) {
+ OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE presentation;
+ InitOMXParams(&presentation);
+ mOMXNode->getParameter(
+ (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAacDrcPresentation,
+ &presentation, sizeof(presentation));
+ int32_t value32 = 0;
+ bool updated = false;
+ if (params->findInt32("aac-pcm-limiter-enable", &value32)) {
+ presentation.nPCMLimiterEnable = value32;
+ updated = true;
+ }
+ if (params->findInt32("aac-encoded-target-level", &value32)) {
+ presentation.nEncodedTargetLevel = value32;
+ updated = true;
+ }
+ if (params->findInt32("aac-drc-cut-level", &value32)) {
+ presentation.nDrcCut = value32;
+ updated = true;
+ }
+ if (params->findInt32("aac-drc-boost-level", &value32)) {
+ presentation.nDrcBoost = value32;
+ updated = true;
+ }
+ if (params->findInt32("aac-drc-heavy-compression", &value32)) {
+ presentation.nHeavyCompression = value32;
+ updated = true;
+ }
+ if (params->findInt32("aac-target-ref-level", &value32)) {
+ presentation.nTargetReferenceLevel = value32;
+ updated = true;
+ }
+ if (params->findInt32("aac-drc-effect-type", &value32)) {
+ presentation.nDrcEffectType = value32;
+ updated = true;
+ }
+ if (params->findInt32("aac-drc-album-mode", &value32)) {
+ presentation.nDrcAlbumMode = value32;
+ updated = true;
+ }
+ if (!params->findInt32("aac-drc-output-loudness", &value32)) {
+ presentation.nDrcOutputLoudness = value32;
+ updated = true;
+ }
+ if (updated) {
+ mOMXNode->setParameter((OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAacDrcPresentation,
+ &presentation, sizeof(presentation));
+ }
+ }
return setVendorParameters(params);
}
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index d67874f..c005bf6 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -2136,8 +2136,10 @@
}
info->sample_rate = srate;
- int32_t cmask = 0;
- if (!meta->findInt32(kKeyChannelMask, &cmask) || cmask == CHANNEL_MASK_USE_CHANNEL_ORDER) {
+ int32_t rawChannelMask;
+ audio_channel_mask_t cmask = meta->findInt32(kKeyChannelMask, &rawChannelMask) ?
+ static_cast<audio_channel_mask_t>(rawChannelMask) : CHANNEL_MASK_USE_CHANNEL_ORDER;
+ if (cmask == CHANNEL_MASK_USE_CHANNEL_ORDER) {
ALOGV("track of type '%s' does not publish channel mask", mime);
// Try a channel count instead
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 2aeddd7..28a7a1e 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -38,6 +38,7 @@
#define DRC_DEFAULT_MOBILE_DRC_HEAVY 1 /* switch for heavy compression for mobile conf */
#define DRC_DEFAULT_MOBILE_DRC_EFFECT 3 /* MPEG-D DRC effect type; 3 => Limited playback range */
#define DRC_DEFAULT_MOBILE_DRC_ALBUM 0 /* MPEG-D DRC album mode; 0 => album mode is disabled, 1 => album mode is enabled */
+#define DRC_DEFAULT_MOBILE_OUTPUT_LOUDNESS -1 /* decoder output loudness; -1 => the value is unknown, otherwise dB step value (e.g. 64 for -16 dB) */
#define DRC_DEFAULT_MOBILE_ENC_LEVEL (-1) /* encoder target level; -1 => the value is unknown, otherwise dB step value (e.g. 64 for -16 dB) */
#define MAX_CHANNEL_COUNT 8 /* maximum number of audio channels that can be decoded */
// names of properties that can be used to override the default DRC settings
@@ -230,6 +231,15 @@
// For seven and eight channel input streams, enable 6.1 and 7.1 channel output
aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, -1);
+ mDrcCompressMode = DRC_DEFAULT_MOBILE_DRC_HEAVY;
+ mDrcTargetRefLevel = DRC_DEFAULT_MOBILE_REF_LEVEL;
+ mDrcEncTargetLevel = DRC_DEFAULT_MOBILE_ENC_LEVEL;
+ mDrcBoostFactor = DRC_DEFAULT_MOBILE_DRC_BOOST;
+ mDrcAttenuationFactor = DRC_DEFAULT_MOBILE_DRC_CUT;
+ mDrcEffectType = DRC_DEFAULT_MOBILE_DRC_EFFECT;
+ mDrcAlbumMode = DRC_DEFAULT_MOBILE_DRC_ALBUM;
+ mDrcOutputLoudness = DRC_DEFAULT_MOBILE_OUTPUT_LOUDNESS;
+
return status;
}
@@ -358,6 +368,27 @@
return OMX_ErrorNone;
}
+ case OMX_IndexParamAudioAndroidAacDrcPresentation:
+ {
+ OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE *aacPresParams =
+ (OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE *)params;
+
+ ALOGD("get OMX_IndexParamAudioAndroidAacDrcPresentation");
+
+ if (!isValidOMXParam(aacPresParams)) {
+ return OMX_ErrorBadParameter;
+ }
+ aacPresParams->nDrcEffectType = mDrcEffectType;
+ aacPresParams->nDrcAlbumMode = mDrcAlbumMode;
+ aacPresParams->nDrcBoost = mDrcBoostFactor;
+ aacPresParams->nDrcCut = mDrcAttenuationFactor;
+ aacPresParams->nHeavyCompression = mDrcCompressMode;
+ aacPresParams->nTargetReferenceLevel = mDrcTargetRefLevel;
+ aacPresParams->nEncodedTargetLevel = mDrcEncTargetLevel;
+ aacPresParams ->nDrcOutputLoudness = mDrcOutputLoudness;
+ return OMX_ErrorNone;
+ }
+
default:
return SimpleSoftOMXComponent::internalGetParameter(index, params);
}
@@ -464,11 +495,13 @@
if (aacPresParams->nDrcEffectType >= -1) {
ALOGV("set nDrcEffectType=%d", aacPresParams->nDrcEffectType);
aacDecoder_SetParam(mAACDecoder, AAC_UNIDRC_SET_EFFECT, aacPresParams->nDrcEffectType);
+ mDrcEffectType = aacPresParams->nDrcEffectType;
}
if (aacPresParams->nDrcAlbumMode >= -1) {
ALOGV("set nDrcAlbumMode=%d", aacPresParams->nDrcAlbumMode);
aacDecoder_SetParam(mAACDecoder, AAC_UNIDRC_ALBUM_MODE,
aacPresParams->nDrcAlbumMode);
+ mDrcAlbumMode = aacPresParams->nDrcAlbumMode;
}
bool updateDrcWrapper = false;
if (aacPresParams->nDrcBoost >= 0) {
@@ -476,34 +509,42 @@
mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR,
aacPresParams->nDrcBoost);
updateDrcWrapper = true;
+ mDrcBoostFactor = aacPresParams->nDrcBoost;
}
if (aacPresParams->nDrcCut >= 0) {
ALOGV("set nDrcCut=%d", aacPresParams->nDrcCut);
mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, aacPresParams->nDrcCut);
updateDrcWrapper = true;
+ mDrcAttenuationFactor = aacPresParams->nDrcCut;
}
if (aacPresParams->nHeavyCompression >= 0) {
ALOGV("set nHeavyCompression=%d", aacPresParams->nHeavyCompression);
mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY,
aacPresParams->nHeavyCompression);
updateDrcWrapper = true;
+ mDrcCompressMode = aacPresParams->nHeavyCompression;
}
if (aacPresParams->nTargetReferenceLevel >= -1) {
ALOGV("set nTargetReferenceLevel=%d", aacPresParams->nTargetReferenceLevel);
mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET,
aacPresParams->nTargetReferenceLevel);
updateDrcWrapper = true;
+ mDrcTargetRefLevel = aacPresParams->nTargetReferenceLevel;
}
if (aacPresParams->nEncodedTargetLevel >= 0) {
ALOGV("set nEncodedTargetLevel=%d", aacPresParams->nEncodedTargetLevel);
mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET,
aacPresParams->nEncodedTargetLevel);
updateDrcWrapper = true;
+ mDrcEncTargetLevel = aacPresParams->nEncodedTargetLevel;
}
if (aacPresParams->nPCMLimiterEnable >= 0) {
aacDecoder_SetParam(mAACDecoder, AAC_PCM_LIMITER_ENABLE,
(aacPresParams->nPCMLimiterEnable != 0));
}
+ if (aacPresParams ->nDrcOutputLoudness != DRC_DEFAULT_MOBILE_OUTPUT_LOUDNESS) {
+ mDrcOutputLoudness = aacPresParams ->nDrcOutputLoudness;
+ }
if (updateDrcWrapper) {
mDrcWrap.update();
}
@@ -854,6 +895,11 @@
// fall through
}
+ if ( mDrcOutputLoudness != mStreamInfo->outputLoudness) {
+ ALOGD("update Loudness, before = %d, now = %d", mDrcOutputLoudness, mStreamInfo->outputLoudness);
+ mDrcOutputLoudness = mStreamInfo->outputLoudness;
+ }
+
/*
* AAC+/eAAC+ streams can be signalled in two ways: either explicitly
* or implicitly, according to MPEG4 spec. AAC+/eAAC+ is a dual
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.h b/media/libstagefright/codecs/aacdec/SoftAAC2.h
index 5bee710..9f98aa1 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.h
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.h
@@ -85,6 +85,17 @@
int32_t mOutputDelayRingBufferWritePos;
int32_t mOutputDelayRingBufferReadPos;
int32_t mOutputDelayRingBufferFilled;
+
+ //drc
+ int32_t mDrcCompressMode;
+ int32_t mDrcTargetRefLevel;
+ int32_t mDrcEncTargetLevel;
+ int32_t mDrcBoostFactor;
+ int32_t mDrcAttenuationFactor;
+ int32_t mDrcEffectType;
+ int32_t mDrcAlbumMode;
+ int32_t mDrcOutputLoudness;
+
bool outputDelayRingBufferPutSamples(INT_PCM *samples, int numSamples);
int32_t outputDelayRingBufferGetSamples(INT_PCM *samples, int numSamples);
int32_t outputDelayRingBufferSamplesAvailable();
diff --git a/media/libstagefright/codecs/m4v_h263/fuzzer/mpeg4_h263_enc_fuzzer.cpp b/media/libstagefright/codecs/m4v_h263/fuzzer/mpeg4_h263_enc_fuzzer.cpp
index f154706..423325d 100644
--- a/media/libstagefright/codecs/m4v_h263/fuzzer/mpeg4_h263_enc_fuzzer.cpp
+++ b/media/libstagefright/codecs/m4v_h263/fuzzer/mpeg4_h263_enc_fuzzer.cpp
@@ -137,7 +137,8 @@
void Codec::encodeFrames(const uint8_t *data, size_t size) {
size_t inputBufferSize = (mFrameWidth * mFrameHeight * 3) / 2;
size_t outputBufferSize = inputBufferSize * 2;
- uint8_t outputBuffer[outputBufferSize];
+ uint8_t *outputBuffer = new uint8_t[outputBufferSize];
+ uint8_t *inputBuffer = new uint8_t[inputBufferSize];
// Get VOL header.
int32_t sizeOutputBuffer = outputBufferSize;
@@ -146,10 +147,9 @@
size_t numFrame = 0;
while (size > 0) {
size_t bytesConsumed = std::min(size, inputBufferSize);
- uint8_t inputBuffer[inputBufferSize];
memcpy(inputBuffer, data, bytesConsumed);
- if (bytesConsumed < sizeof(inputBuffer)) {
- memset(inputBuffer + bytesConsumed, data[0], sizeof(inputBuffer) - bytesConsumed);
+ if (bytesConsumed < inputBufferSize) {
+ memset(inputBuffer + bytesConsumed, data[0], inputBufferSize - bytesConsumed);
}
VideoEncFrameIO videoIn{}, videoOut{};
videoIn.height = mFrameHeight;
@@ -170,6 +170,8 @@
data += bytesConsumed;
size -= bytesConsumed;
}
+ delete[] inputBuffer;
+ delete[] outputBuffer;
}
extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {
diff --git a/media/libstagefright/omx/SimpleSoftOMXComponent.cpp b/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
index ddb459f..44415aa 100644
--- a/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
+++ b/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
@@ -17,6 +17,10 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "SimpleSoftOMXComponent"
#include <utils/Log.h>
+#include <OMX_Core.h>
+#include <OMX_Audio.h>
+#include <OMX_IndexExt.h>
+#include <OMX_AudioExt.h>
#include <media/stagefright/omx/SimpleSoftOMXComponent.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -74,7 +78,7 @@
OMX_U32 portIndex;
- switch (index) {
+ switch ((int)index) {
case OMX_IndexParamPortDefinition:
{
const OMX_PARAM_PORTDEFINITIONTYPE *portDefs =
@@ -108,6 +112,19 @@
break;
}
+ case OMX_IndexParamAudioAndroidAacDrcPresentation:
+ {
+ if (mState == OMX_StateInvalid) {
+ return false;
+ }
+ const OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE *aacPresParams =
+ (const OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE *)params;
+ if (!isValidOMXParam(aacPresParams)) {
+ return false;
+ }
+ return true;
+ }
+
default:
return false;
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 20f561e..c47afd5 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -405,7 +405,7 @@
case AUDIO_CHANNEL_REPRESENTATION_POSITION: {
// Haptic channel mask is only applicable for channel position mask.
const uint32_t channelCount = audio_channel_count_from_out_mask(
- channelMask & ~AUDIO_CHANNEL_HAPTIC_ALL);
+ static_cast<audio_channel_mask_t>(channelMask & ~AUDIO_CHANNEL_HAPTIC_ALL));
const uint32_t maxChannelCount = kEnableExtendedChannels
? AudioMixer::MAX_NUM_CHANNELS : FCC_2;
if (channelCount < FCC_2 // mono is not supported at this time
diff --git a/services/audioflinger/SpdifStreamOut.cpp b/services/audioflinger/SpdifStreamOut.cpp
index c7aba79..0ce5681 100644
--- a/services/audioflinger/SpdifStreamOut.cpp
+++ b/services/audioflinger/SpdifStreamOut.cpp
@@ -39,7 +39,7 @@
, mSpdifEncoder(this, format)
, mApplicationFormat(AUDIO_FORMAT_DEFAULT)
, mApplicationSampleRate(0)
- , mApplicationChannelMask(0)
+ , mApplicationChannelMask(AUDIO_CHANNEL_NONE)
{
}
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index b143388..00fbbb0 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2863,8 +2863,8 @@
(void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize);
}
- mHapticChannelMask = mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL;
- mChannelMask &= ~mHapticChannelMask;
+ mHapticChannelMask = static_cast<audio_channel_mask_t>(mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL);
+ mChannelMask = static_cast<audio_channel_mask_t>(mChannelMask & ~mHapticChannelMask);
mHapticChannelCount = audio_channel_count_from_out_mask(mHapticChannelMask);
mChannelCount -= mHapticChannelCount;
@@ -4200,7 +4200,7 @@
"Enumerated device type(%#x) must not be used "
"as it does not support audio patches",
patch->sinks[i].ext.device.type);
- type |= patch->sinks[i].ext.device.type;
+ type = static_cast<audio_devices_t>(type | patch->sinks[i].ext.device.type);
deviceTypeAddrs.push_back(AudioDeviceTypeAddr(patch->sinks[i].ext.device.type,
patch->sinks[i].ext.device.address));
}
@@ -4450,8 +4450,9 @@
// wrap the source side of the MonoPipe to make it an AudioBufferProvider
fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
fastTrack->mVolumeProvider = NULL;
- fastTrack->mChannelMask = mChannelMask | mHapticChannelMask; // mPipeSink channel mask for
- // audio to FastMixer
+ fastTrack->mChannelMask = static_cast<audio_channel_mask_t>(
+ mChannelMask | mHapticChannelMask); // mPipeSink channel mask for
+ // audio to FastMixer
fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
fastTrack->mHapticPlaybackEnabled = mHapticChannelMask != AUDIO_CHANNEL_NONE;
fastTrack->mHapticIntensity = AudioMixer::HAPTIC_SCALE_NONE;
@@ -4465,7 +4466,8 @@
// specify sink channel mask when haptic channel mask present as it can not
// be calculated directly from channel count
state->mSinkChannelMask = mHapticChannelMask == AUDIO_CHANNEL_NONE
- ? AUDIO_CHANNEL_NONE : mChannelMask | mHapticChannelMask;
+ ? AUDIO_CHANNEL_NONE
+ : static_cast<audio_channel_mask_t>(mChannelMask | mHapticChannelMask);
state->mCommand = FastMixerState::COLD_IDLE;
// already done in constructor initialization list
//mFastMixerFutex = 0;
@@ -9185,7 +9187,7 @@
"Enumerated device type(%#x) must not be used "
"as it does not support audio patches",
patch->sinks[i].ext.device.type);
- type |= patch->sinks[i].ext.device.type;
+ type = static_cast<audio_devices_t>(type | patch->sinks[i].ext.device.type);
sinkDeviceTypeAddrs.push_back(AudioDeviceTypeAddr(patch->sinks[i].ext.device.type,
patch->sinks[i].ext.device.address));
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index 5120aeb..3d97440 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -380,7 +380,7 @@
if (isEmpty()) {
// Return nullptr if this collection is empty.
return nullptr;
- } else if (areAllOfSameDeviceType(types(), audio_is_input_device)) {
+ } else if (areAllOfSameDeviceType(types(), audio_call_is_input_device)) {
// For input case, return the first one when there is only one device.
return size() > 1 ? nullptr : *begin();
} else if (areAllOfSameDeviceType(types(), audio_is_output_device)) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
index 883e713..889f031 100644
--- a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
@@ -337,7 +337,7 @@
std::string mode = getXmlAttribute(cur, Attributes::mode);
if (!mode.empty()) {
- gain->setMode(GainModeConverter::maskFromString(mode));
+ gain->setMode(static_cast<audio_gain_mode_t>(GainModeConverter::maskFromString(mode)));
}
std::string channelsLiteral = getXmlAttribute(cur, Attributes::channelMask);
diff --git a/services/audiopolicy/config/a2dp_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/a2dp_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..2d323f6
--- /dev/null
+++ b/services/audiopolicy/config/a2dp_audio_policy_configuration_7_0.xml
@@ -0,0 +1,44 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- A2dp Audio HAL Audio Policy Configuration file -->
+<module name="a2dp" halVersion="2.0">
+ <mixPorts>
+ <mixPort name="a2dp output" role="source"/>
+ <mixPort name="a2dp input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP In" type="AUDIO_DEVICE_IN_BLUETOOTH_A2DP" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="BT A2DP Out"
+ sources="a2dp output"/>
+ <route type="mix" sink="BT A2DP Headphones"
+ sources="a2dp output"/>
+ <route type="mix" sink="BT A2DP Speaker"
+ sources="a2dp output"/>
+ <route type="mix" sink="a2dp input"
+ sources="BT A2DP In"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/config/a2dp_in_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/a2dp_in_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..d59ad70
--- /dev/null
+++ b/services/audiopolicy/config/a2dp_in_audio_policy_configuration_7_0.xml
@@ -0,0 +1,22 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Bluetooth Input Audio HAL Audio Policy Configuration file -->
+<module name="a2dp" halVersion="2.0">
+ <mixPorts>
+ <mixPort name="a2dp input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="BT A2DP In" type="AUDIO_DEVICE_IN_BLUETOOTH_A2DP" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="a2dp input"
+ sources="BT A2DP In"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/config/audio_policy_configuration_7_0.xml b/services/audiopolicy/config/audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..b30ab30
--- /dev/null
+++ b/services/audiopolicy/config/audio_policy_configuration_7_0.xml
@@ -0,0 +1,211 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2019 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <!-- version section contains a “version” tag in the form “major.minor” e.g version=”1.0” -->
+
+ <!-- Global configuration Decalaration -->
+ <globalConfiguration speaker_drc_enabled="true"/>
+
+
+ <!-- Modules section:
+ There is one section per audio HW module present on the platform.
+ Each module section will contains two mandatory tags for audio HAL “halVersion” and “name”.
+ The module names are the same as in current .conf file:
+ “primary”, “A2DP”, “remote_submix”, “USB”
+ Each module will contain the following sections:
+ “devicePorts”: a list of device descriptors for all input and output devices accessible via this
+ module.
+ This contains both permanently attached devices and removable devices.
+ “mixPorts”: listing all output and input streams exposed by the audio HAL
+ “routes”: list of possible connections between input and output devices or between stream and
+ devices.
+ "route": is defined by an attribute:
+ -"type": <mux|mix> means all sources are mutual exclusive (mux) or can be mixed (mix)
+ -"sink": the sink involved in this route
+ -"sources": all the sources than can be connected to the sink via vis route
+ “attachedDevices”: permanently attached devices.
+ The attachedDevices section is a list of devices names. The names correspond to device names
+ defined in <devicePorts> section.
+ “defaultOutputDevice”: device to be used by default when no policy rule applies
+ -->
+ <modules>
+ <!-- Primary Audio HAL -->
+ <module name="primary" halVersion="3.0">
+ <attachedDevices>
+ <item>Speaker</item>
+ <item>Built-In Mic</item>
+ <item>Built-In Back Mic</item>
+ </attachedDevices>
+ <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <mixPorts>
+ <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="deep_buffer" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DEEP_BUFFER">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="compressed_offload" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD AUDIO_OUTPUT_FLAG_NON_BLOCKING">
+ <profile name="" format="AUDIO_FORMAT_MP3"
+ samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+ <profile name="" format="AUDIO_FORMAT_AAC"
+ samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+ <profile name="" format="AUDIO_FORMAT_AAC_LC"
+ samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+ </mixPort>
+ <mixPort name="voice_tx" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ </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="voice_rx" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <!-- Output devices declaration, i.e. Sink DEVICE PORT -->
+ <devicePort tagName="Earpiece" type="AUDIO_DEVICE_OUT_EARPIECE" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </devicePort>
+ <devicePort tagName="Speaker" role="sink" type="AUDIO_DEVICE_OUT_SPEAKER" address="">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="gain_1" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-8400"
+ maxValueMB="4000"
+ defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="Wired Headset" type="AUDIO_DEVICE_OUT_WIRED_HEADSET" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="Wired Headphones" type="AUDIO_DEVICE_OUT_WIRED_HEADPHONE" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ </devicePort>
+ <devicePort tagName="BT SCO Headset" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ </devicePort>
+ <devicePort tagName="BT SCO Car Kit" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ </devicePort>
+ <devicePort tagName="Telephony Tx" type="AUDIO_DEVICE_OUT_TELEPHONY_TX" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ </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="Wired Headset Mic" type="AUDIO_DEVICE_IN_WIRED_HEADSET" 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="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </devicePort>
+ <devicePort tagName="Telephony Rx" type="AUDIO_DEVICE_IN_TELEPHONY_RX" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </devicePort>
+ </devicePorts>
+ <!-- route declaration, i.e. list all available sources for a given sink -->
+ <routes>
+ <route type="mix" sink="Earpiece"
+ sources="primary output,deep_buffer,BT SCO Headset Mic"/>
+ <route type="mix" sink="Speaker"
+ sources="primary output,deep_buffer,compressed_offload,BT SCO Headset Mic,Telephony Rx"/>
+ <route type="mix" sink="Wired Headset"
+ sources="primary output,deep_buffer,compressed_offload,BT SCO Headset Mic,Telephony Rx"/>
+ <route type="mix" sink="Wired Headphones"
+ sources="primary output,deep_buffer,compressed_offload,BT SCO Headset Mic,Telephony Rx"/>
+ <route type="mix" sink="primary input"
+ sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
+ <route type="mix" sink="Telephony Tx"
+ sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic, voice_tx"/>
+ <route type="mix" sink="voice_rx"
+ sources="Telephony Rx"/>
+ </routes>
+
+ </module>
+
+ <!-- A2dp Input Audio HAL -->
+ <xi:include href="a2dp_in_audio_policy_configuration_7_0.xml"/>
+
+ <!-- Usb Audio HAL -->
+ <xi:include href="usb_audio_policy_configuration.xml"/>
+
+ <!-- Remote Submix Audio HAL -->
+ <xi:include href="r_submix_audio_policy_configuration.xml"/>
+
+ <!-- Bluetooth Audio HAL -->
+ <xi:include href="bluetooth_audio_policy_configuration_7_0.xml"/>
+
+ <!-- MSD Audio HAL (optional) -->
+ <xi:include href="msd_audio_policy_configuration_7_0.xml"/>
+
+ </modules>
+ <!-- End of Modules section -->
+
+ <!-- Volume section:
+ IMPORTANT NOTE: Volume tables have been moved to engine configuration.
+ Keep it here for legacy.
+ Engine will fallback on these files if none are provided by engine.
+ -->
+
+ <xi:include href="audio_policy_volumes.xml"/>
+ <xi:include href="default_volume_tables.xml"/>
+
+ <!-- End of Volume section -->
+
+ <!-- Surround Sound configuration -->
+
+ <xi:include href="surround_sound_configuration_5_0.xml"/>
+
+ <!-- End of Surround Sound configuration -->
+
+</audioPolicyConfiguration>
diff --git a/services/audiopolicy/config/bluetooth_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/bluetooth_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..2dffe02
--- /dev/null
+++ b/services/audiopolicy/config/bluetooth_audio_policy_configuration_7_0.xml
@@ -0,0 +1,44 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Bluetooth Audio HAL Audio Policy Configuration file -->
+<module name="bluetooth" halVersion="2.0">
+ <mixPorts>
+ <!-- A2DP Audio Ports -->
+ <mixPort name="a2dp output" role="source"/>
+ <!-- Hearing AIDs Audio Ports -->
+ <mixPort name="hearing aid output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="24000 16000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <!-- A2DP Audio Ports -->
+ <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000 88200 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000 88200 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000 88200 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <!-- Hearing AIDs Audio Ports -->
+ <devicePort tagName="BT Hearing Aid Out" type="AUDIO_DEVICE_OUT_HEARING_AID" role="sink"/>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="BT A2DP Out"
+ sources="a2dp output"/>
+ <route type="mix" sink="BT A2DP Headphones"
+ sources="a2dp output"/>
+ <route type="mix" sink="BT A2DP Speaker"
+ sources="a2dp output"/>
+ <route type="mix" sink="BT Hearing Aid Out"
+ sources="hearing aid output"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/config/hearing_aid_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/hearing_aid_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..8c364e4
--- /dev/null
+++ b/services/audiopolicy/config/hearing_aid_audio_policy_configuration_7_0.xml
@@ -0,0 +1,17 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Hearing aid Audio HAL Audio Policy Configuration file -->
+<module name="hearing_aid" halVersion="2.0">
+ <mixPorts>
+ <mixPort name="hearing aid output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="24000 16000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="BT Hearing Aid Out" type="AUDIO_DEVICE_OUT_HEARING_AID" role="sink"/>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="BT Hearing Aid Out" sources="hearing aid output"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/config/msd_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/msd_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..f167f0b
--- /dev/null
+++ b/services/audiopolicy/config/msd_audio_policy_configuration_7_0.xml
@@ -0,0 +1,78 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Copyright (C) 2017-2018 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.
+-->
+<!-- Multi Stream Decoder Audio Policy Configuration file -->
+<module name="msd" halVersion="2.0">
+ <attachedDevices>
+ <item>MS12 Input</item>
+ <item>MS12 Output</item>
+ </attachedDevices>
+ <mixPorts>
+ <mixPort name="ms12 input" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="ms12 compressed input" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD AUDIO_OUTPUT_FLAG_NON_BLOCKING">
+ <profile name="" format="AUDIO_FORMAT_AC3"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3_JOC"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_AC4"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ </mixPort>
+ <!-- The HW AV Sync flag is not required, but is recommended -->
+ <mixPort name="ms12 output" role="sink" flags="AUDIO_INPUT_FLAG_HW_AV_SYNC AUDIO_INPUT_FLAG_DIRECT">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ <profile name="" format="AUDIO_FORMAT_AC3"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_5POINT1"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="MS12 Input" type="AUDIO_DEVICE_OUT_BUS" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <profile name="" format="AUDIO_FORMAT_AC3"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3_JOC"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_AC4"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ </devicePort>
+ <devicePort tagName="MS12 Output" type="AUDIO_DEVICE_IN_BUS" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="MS12 Input" sources="ms12 input,ms12 compressed input"/>
+ <route type="mix" sink="ms12 output" sources="MS12 Output"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/config/primary_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/primary_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..68a56b2
--- /dev/null
+++ b/services/audiopolicy/config/primary_audio_policy_configuration_7_0.xml
@@ -0,0 +1,32 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Default Primary Audio HAL Module Audio Policy Configuration include file -->
+<module name="primary" halVersion="2.0">
+ <attachedDevices>
+ <item>Speaker</item>
+ <item>Built-In Mic</item>
+ </attachedDevices>
+ <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <mixPorts>
+ <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="primary input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+ </devicePort>
+
+ <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="Speaker"
+ sources="primary output"/>
+ <route type="mix" sink="primary input"
+ sources="Built-In Mic"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/engine/common/src/EngineDefaultConfig.h b/services/audiopolicy/engine/common/src/EngineDefaultConfig.h
index 1821140..d39eff6 100644
--- a/services/audiopolicy/engine/common/src/EngineDefaultConfig.h
+++ b/services/audiopolicy/engine/common/src/EngineDefaultConfig.h
@@ -26,8 +26,8 @@
{"STRATEGY_PHONE",
{
{"phone", AUDIO_STREAM_VOICE_CALL, "AUDIO_STREAM_VOICE_CALL",
- {{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_VOICE_COMMUNICATION, AUDIO_SOURCE_DEFAULT, 0,
- ""}},
+ {{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_VOICE_COMMUNICATION, AUDIO_SOURCE_DEFAULT,
+ AUDIO_FLAG_NONE, ""}},
},
{"sco", AUDIO_STREAM_BLUETOOTH_SCO, "AUDIO_STREAM_BLUETOOTH_SCO",
{{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_SCO,
@@ -39,10 +39,11 @@
{
{"ring", AUDIO_STREAM_RING, "AUDIO_STREAM_RING",
{{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE,
- AUDIO_SOURCE_DEFAULT, 0, ""}}
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""}}
},
{"alarm", AUDIO_STREAM_ALARM, "AUDIO_STREAM_ALARM",
- {{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_ALARM, AUDIO_SOURCE_DEFAULT, 0, ""}},
+ {{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_ALARM, AUDIO_SOURCE_DEFAULT,
+ AUDIO_FLAG_NONE, ""}},
}
},
},
@@ -58,7 +59,7 @@
{
{"", AUDIO_STREAM_ACCESSIBILITY, "AUDIO_STREAM_ACCESSIBILITY",
{{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY,
- AUDIO_SOURCE_DEFAULT, 0, ""}}
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""}}
}
},
},
@@ -66,15 +67,16 @@
{
{"", AUDIO_STREAM_NOTIFICATION, "AUDIO_STREAM_NOTIFICATION",
{
- {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION, AUDIO_SOURCE_DEFAULT, 0, ""},
+ {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION, AUDIO_SOURCE_DEFAULT,
+ AUDIO_FLAG_NONE, ""},
{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_NOTIFICATION_EVENT,
- AUDIO_SOURCE_DEFAULT, 0, ""}
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""}
}
}
},
@@ -83,21 +85,25 @@
{
{"assistant", AUDIO_STREAM_ASSISTANT, "AUDIO_STREAM_ASSISTANT",
{{AUDIO_CONTENT_TYPE_SPEECH, AUDIO_USAGE_ASSISTANT,
- AUDIO_SOURCE_DEFAULT, 0, ""}}
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""}}
},
{"music", AUDIO_STREAM_MUSIC, "AUDIO_STREAM_MUSIC",
{
- {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_MEDIA, AUDIO_SOURCE_DEFAULT, 0, ""},
- {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_GAME, AUDIO_SOURCE_DEFAULT, 0, ""},
- {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_ASSISTANT, AUDIO_SOURCE_DEFAULT, 0, ""},
+ {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_MEDIA, AUDIO_SOURCE_DEFAULT,
+ AUDIO_FLAG_NONE, ""},
+ {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_GAME, AUDIO_SOURCE_DEFAULT,
+ AUDIO_FLAG_NONE, ""},
+ {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_ASSISTANT, AUDIO_SOURCE_DEFAULT,
+ AUDIO_FLAG_NONE, ""},
{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE,
- AUDIO_SOURCE_DEFAULT, 0, ""},
- {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_DEFAULT, 0, ""}
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
+ {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_DEFAULT,
+ AUDIO_FLAG_NONE, ""}
},
},
{"system", AUDIO_STREAM_SYSTEM, "AUDIO_STREAM_SYSTEM",
{{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_ASSISTANCE_SONIFICATION,
- AUDIO_SOURCE_DEFAULT, 0, ""}}
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""}}
}
},
},
@@ -106,7 +112,7 @@
{"", AUDIO_STREAM_DTMF, "AUDIO_STREAM_DTMF",
{
{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
- AUDIO_SOURCE_DEFAULT, 0, ""}
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""}
}
}
},
@@ -114,7 +120,8 @@
{"STRATEGY_CALL_ASSISTANT",
{
{"", AUDIO_STREAM_CALL_ASSISTANT, "AUDIO_STREAM_CALL_ASSISTANT",
- {{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_CALL_ASSISTANT, AUDIO_SOURCE_DEFAULT, 0, ""}}
+ {{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_CALL_ASSISTANT, AUDIO_SOURCE_DEFAULT,
+ AUDIO_FLAG_NONE, ""}}
}
},
},
@@ -136,14 +143,16 @@
{"rerouting",
{
{"", AUDIO_STREAM_REROUTING, "AUDIO_STREAM_REROUTING",
- {{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_VIRTUAL_SOURCE, AUDIO_SOURCE_DEFAULT, 0, ""}}
+ {{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_VIRTUAL_SOURCE, AUDIO_SOURCE_DEFAULT,
+ AUDIO_FLAG_NONE, ""}}
}
},
},
{"patch",
{
{"", AUDIO_STREAM_PATCH, "AUDIO_STREAM_PATCH",
- {{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_DEFAULT, 0, ""}}
+ {{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_DEFAULT,
+ AUDIO_FLAG_NONE, ""}}
}
},
}
diff --git a/services/audiopolicy/engine/config/src/EngineConfig.cpp b/services/audiopolicy/engine/config/src/EngineConfig.cpp
index 4842cb2..daf6418 100644
--- a/services/audiopolicy/engine/config/src/EngineConfig.cpp
+++ b/services/audiopolicy/engine/config/src/EngineConfig.cpp
@@ -228,7 +228,8 @@
std::string flags = getXmlAttribute(cur, "value");
ALOGV("%s flags %s", __FUNCTION__, flags.c_str());
- attributes.flags = AudioFlagConverter::maskFromString(flags, " ");
+ attributes.flags = static_cast<audio_flags_mask_t>(
+ AudioFlagConverter::maskFromString(flags, " "));
}
if (!xmlStrcmp(cur->name, (const xmlChar *)("Bundle"))) {
std::string bundleKey = getXmlAttribute(cur, "key");
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/plugin/InputSource.cpp b/services/audiopolicy/engineconfigurable/parameter-framework/plugin/InputSource.cpp
index f91f8d7..f8a6fc0 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/plugin/InputSource.cpp
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/plugin/InputSource.cpp
@@ -45,7 +45,7 @@
bool InputSource::sendToHW(string & /*error*/)
{
- uint32_t applicableInputDevice;
+ audio_devices_t applicableInputDevice;
blackboardRead(&applicableInputDevice, sizeof(applicableInputDevice));
return mPolicyPluginInterface->setDeviceForInputSource(mId, applicableInputDevice);
}
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/plugin/ProductStrategy.h b/services/audiopolicy/engineconfigurable/parameter-framework/plugin/ProductStrategy.h
index 244f082..6c8eb65 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/plugin/ProductStrategy.h
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/plugin/ProductStrategy.h
@@ -32,7 +32,7 @@
struct Device
{
- uint32_t applicableDevice; /**< applicable device for this strategy. */
+ audio_devices_t applicableDevice; /**< applicable device for this strategy. */
char deviceAddress[mMaxStringSize]; /**< device address associated with this strategy. */
} __attribute__((packed));
diff --git a/services/audiopolicy/engineconfigurable/src/InputSource.cpp b/services/audiopolicy/engineconfigurable/src/InputSource.cpp
index aa06ae3..f4645e6 100644
--- a/services/audiopolicy/engineconfigurable/src/InputSource.cpp
+++ b/services/audiopolicy/engineconfigurable/src/InputSource.cpp
@@ -51,7 +51,7 @@
mApplicableDevices = devices;
return NO_ERROR;
}
- devices |= AUDIO_DEVICE_BIT_IN;
+ devices = static_cast<audio_devices_t>(devices | AUDIO_DEVICE_BIT_IN);
if (!audio_is_input_device(devices)) {
ALOGE("%s: trying to set an invalid device 0x%X for input source %s",
__FUNCTION__, devices, getName().c_str());
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index ae71959..047b804 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -903,7 +903,8 @@
// Only honor audibility enforced when required. The client will be
// forced to reconnect if the forced usage changes.
if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
- dstAttr->flags &= ~AUDIO_FLAG_AUDIBILITY_ENFORCED;
+ dstAttr->flags = static_cast<audio_flags_mask_t>(
+ dstAttr->flags & ~AUDIO_FLAG_AUDIBILITY_ENFORCED);
}
return NO_ERROR;
@@ -935,7 +936,7 @@
return status;
}
if (auto it = mAllowedCapturePolicies.find(uid); it != end(mAllowedCapturePolicies)) {
- resultAttr->flags |= it->second;
+ resultAttr->flags = static_cast<audio_flags_mask_t>(resultAttr->flags | it->second);
}
*stream = mEngine->getStreamTypeForAttributes(*resultAttr);
@@ -1253,7 +1254,8 @@
// Discard haptic channel mask when forcing muting haptic channels.
audio_channel_mask_t channelMask = forceMutingHaptic
- ? (config->channel_mask & ~AUDIO_CHANNEL_HAPTIC_ALL) : config->channel_mask;
+ ? static_cast<audio_channel_mask_t>(config->channel_mask & ~AUDIO_CHANNEL_HAPTIC_ALL)
+ : config->channel_mask;
// open a direct output if required by specified parameters
//force direct flag if offload flag is set: offloading implies a direct output stream
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 34d07b6..8f1a7f7 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -244,11 +244,12 @@
uid = callingUid;
}
if (!mPackageManager.allowPlaybackCapture(uid)) {
- attr->flags |= AUDIO_FLAG_NO_MEDIA_PROJECTION;
+ attr->flags = static_cast<audio_flags_mask_t>(attr->flags | AUDIO_FLAG_NO_MEDIA_PROJECTION);
}
if (((attr->flags & (AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY|AUDIO_FLAG_BYPASS_MUTE)) != 0)
&& !bypassInterruptionPolicyAllowed(pid, uid)) {
- attr->flags &= ~(AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY|AUDIO_FLAG_BYPASS_MUTE);
+ attr->flags = static_cast<audio_flags_mask_t>(
+ attr->flags & ~(AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY|AUDIO_FLAG_BYPASS_MUTE));
}
AutoCallerClear acc;
AudioPolicyInterface::output_type_t outputType;
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index a0074bc..0bd5442 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -87,7 +87,7 @@
void getOutputForAttr(
audio_port_handle_t *selectedDeviceId,
audio_format_t format,
- int channelMask,
+ audio_channel_mask_t channelMask,
int sampleRate,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
audio_io_handle_t *output = nullptr,
@@ -98,7 +98,7 @@
audio_unique_id_t riid,
audio_port_handle_t *selectedDeviceId,
audio_format_t format,
- int channelMask,
+ audio_channel_mask_t channelMask,
int sampleRate,
audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
audio_port_handle_t *portId = nullptr);
@@ -164,7 +164,7 @@
void AudioPolicyManagerTest::getOutputForAttr(
audio_port_handle_t *selectedDeviceId,
audio_format_t format,
- int channelMask,
+ audio_channel_mask_t channelMask,
int sampleRate,
audio_output_flags_t flags,
audio_io_handle_t *output,
@@ -194,7 +194,7 @@
audio_unique_id_t riid,
audio_port_handle_t *selectedDeviceId,
audio_format_t format,
- int channelMask,
+ audio_channel_mask_t channelMask,
int sampleRate,
audio_input_flags_t flags,
audio_port_handle_t *portId) {
@@ -707,7 +707,8 @@
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
audio_source_t source = AUDIO_SOURCE_REMOTE_SUBMIX;
- audio_attributes_t attr = {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, source, 0, ""};
+ audio_attributes_t attr = {
+ AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, source, AUDIO_FLAG_NONE, ""};
std::string tags = "addr=" + mMixAddress;
strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
getInputForAttr(attr, mTracker->getRiid(), &selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT,
@@ -757,9 +758,9 @@
AudioPolicyManagerTestDPPlaybackReRouting,
testing::Values(
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_MEDIA,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ALARM,
- AUDIO_SOURCE_DEFAULT, 0, ""}
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""}
)
);
@@ -768,47 +769,47 @@
AudioPolicyManagerTestDPPlaybackReRouting,
testing::Values(
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_MEDIA,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_VOICE_COMMUNICATION,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
- AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ALARM,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
- AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE,
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
- AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST,
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
- AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT,
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
- AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED,
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION_EVENT,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
- AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY,
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
- AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE,
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
- AUDIO_USAGE_ASSISTANCE_SONIFICATION,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_USAGE_ASSISTANCE_SONIFICATION,
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_GAME,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_VIRTUAL_SOURCE,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ASSISTANT,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_SPEECH, AUDIO_USAGE_ASSISTANT,
- AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"}
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, "addr=remote_submix_media"}
)
);
@@ -817,41 +818,41 @@
AudioPolicyManagerTestDPPlaybackReRouting,
testing::Values(
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_VOICE_COMMUNICATION,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION_EVENT,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
AUDIO_USAGE_ASSISTANCE_SONIFICATION,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_GAME,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ASSISTANT,
- AUDIO_SOURCE_DEFAULT, 0, ""},
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_SPEECH, AUDIO_USAGE_ASSISTANT,
- AUDIO_SOURCE_DEFAULT, 0, ""}
+ AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""}
)
);
@@ -892,7 +893,8 @@
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
audio_usage_t usage = AUDIO_USAGE_VIRTUAL_SOURCE;
- audio_attributes_t attr = {AUDIO_CONTENT_TYPE_UNKNOWN, usage, AUDIO_SOURCE_DEFAULT, 0, ""};
+ audio_attributes_t attr =
+ {AUDIO_CONTENT_TYPE_UNKNOWN, usage, AUDIO_SOURCE_DEFAULT, AUDIO_FLAG_NONE, ""};
std::string tags = std::string("addr=") + mMixAddress;
strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
@@ -941,17 +943,19 @@
AudioPolicyManagerTestDPMixRecordInjection,
testing::Values(
(audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
- AUDIO_SOURCE_CAMCORDER, 0, ""},
+ AUDIO_SOURCE_CAMCORDER, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
- AUDIO_SOURCE_CAMCORDER, 0, "addr=remote_submix_media"},
+ AUDIO_SOURCE_CAMCORDER, AUDIO_FLAG_NONE,
+ "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
- AUDIO_SOURCE_MIC, 0, "addr=remote_submix_media"},
+ AUDIO_SOURCE_MIC, AUDIO_FLAG_NONE,
+ "addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
- AUDIO_SOURCE_MIC, 0, ""},
+ AUDIO_SOURCE_MIC, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
- AUDIO_SOURCE_VOICE_COMMUNICATION, 0, ""},
+ AUDIO_SOURCE_VOICE_COMMUNICATION, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
- AUDIO_SOURCE_VOICE_COMMUNICATION, 0,
+ AUDIO_SOURCE_VOICE_COMMUNICATION, AUDIO_FLAG_NONE,
"addr=remote_submix_media"}
)
);
@@ -962,14 +966,15 @@
AudioPolicyManagerTestDPMixRecordInjection,
testing::Values(
(audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
- AUDIO_SOURCE_VOICE_RECOGNITION, 0, ""},
+ AUDIO_SOURCE_VOICE_RECOGNITION, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
- AUDIO_SOURCE_HOTWORD, 0, ""},
+ AUDIO_SOURCE_HOTWORD, AUDIO_FLAG_NONE, ""},
(audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
- AUDIO_SOURCE_VOICE_RECOGNITION, 0,
+ AUDIO_SOURCE_VOICE_RECOGNITION, AUDIO_FLAG_NONE,
"addr=remote_submix_media"},
(audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
- AUDIO_SOURCE_HOTWORD, 0, "addr=remote_submix_media"}
+ AUDIO_SOURCE_HOTWORD, AUDIO_FLAG_NONE,
+ "addr=remote_submix_media"}
)
);
diff --git a/services/mediametrics/Android.bp b/services/mediametrics/Android.bp
index f033d5c..67e6c39 100644
--- a/services/mediametrics/Android.bp
+++ b/services/mediametrics/Android.bp
@@ -111,7 +111,7 @@
],
}
-cc_library_shared {
+cc_library {
name: "libmediametricsservice",
defaults: [
"mediametrics_flags_defaults",
diff --git a/services/mediametrics/statsd_audiopolicy.cpp b/services/mediametrics/statsd_audiopolicy.cpp
index 393c6ae..6ef2f2c 100644
--- a/services/mediametrics/statsd_audiopolicy.cpp
+++ b/services/mediametrics/statsd_audiopolicy.cpp
@@ -32,7 +32,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "frameworks/base/core/proto/android/stats/mediametrics/mediametrics.pb.h"
+#include "frameworks/proto_logging/stats/enums/stats/mediametrics/mediametrics.pb.h"
#include "iface_statsd.h"
namespace android {
diff --git a/services/mediametrics/statsd_audiorecord.cpp b/services/mediametrics/statsd_audiorecord.cpp
index 43feda1..76f4b59 100644
--- a/services/mediametrics/statsd_audiorecord.cpp
+++ b/services/mediametrics/statsd_audiorecord.cpp
@@ -32,7 +32,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "frameworks/base/core/proto/android/stats/mediametrics/mediametrics.pb.h"
+#include "frameworks/proto_logging/stats/enums/stats/mediametrics/mediametrics.pb.h"
#include "iface_statsd.h"
namespace android {
diff --git a/services/mediametrics/statsd_audiothread.cpp b/services/mediametrics/statsd_audiothread.cpp
index e867f5b..2ad2562 100644
--- a/services/mediametrics/statsd_audiothread.cpp
+++ b/services/mediametrics/statsd_audiothread.cpp
@@ -32,7 +32,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "frameworks/base/core/proto/android/stats/mediametrics/mediametrics.pb.h"
+#include "frameworks/proto_logging/stats/enums/stats/mediametrics/mediametrics.pb.h"
#include "iface_statsd.h"
namespace android {
diff --git a/services/mediametrics/statsd_audiotrack.cpp b/services/mediametrics/statsd_audiotrack.cpp
index ee5b9b2..6b08a78 100644
--- a/services/mediametrics/statsd_audiotrack.cpp
+++ b/services/mediametrics/statsd_audiotrack.cpp
@@ -32,7 +32,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "frameworks/base/core/proto/android/stats/mediametrics/mediametrics.pb.h"
+#include "frameworks/proto_logging/stats/enums/stats/mediametrics/mediametrics.pb.h"
#include "iface_statsd.h"
namespace android {
diff --git a/services/mediametrics/statsd_codec.cpp b/services/mediametrics/statsd_codec.cpp
index ec9354f..d502b30 100644
--- a/services/mediametrics/statsd_codec.cpp
+++ b/services/mediametrics/statsd_codec.cpp
@@ -33,7 +33,7 @@
#include "cleaner.h"
#include "MediaMetricsService.h"
-#include "frameworks/base/core/proto/android/stats/mediametrics/mediametrics.pb.h"
+#include "frameworks/proto_logging/stats/enums/stats/mediametrics/mediametrics.pb.h"
#include "iface_statsd.h"
namespace android {
diff --git a/services/mediametrics/statsd_extractor.cpp b/services/mediametrics/statsd_extractor.cpp
index 3d5739f..16814d9 100644
--- a/services/mediametrics/statsd_extractor.cpp
+++ b/services/mediametrics/statsd_extractor.cpp
@@ -32,7 +32,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "frameworks/base/core/proto/android/stats/mediametrics/mediametrics.pb.h"
+#include "frameworks/proto_logging/stats/enums/stats/mediametrics/mediametrics.pb.h"
#include "iface_statsd.h"
namespace android {
diff --git a/services/mediametrics/statsd_nuplayer.cpp b/services/mediametrics/statsd_nuplayer.cpp
index 488bdcb..a8d0f55 100644
--- a/services/mediametrics/statsd_nuplayer.cpp
+++ b/services/mediametrics/statsd_nuplayer.cpp
@@ -32,7 +32,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "frameworks/base/core/proto/android/stats/mediametrics/mediametrics.pb.h"
+#include "frameworks/proto_logging/stats/enums/stats/mediametrics/mediametrics.pb.h"
#include "iface_statsd.h"
namespace android {
diff --git a/services/mediametrics/statsd_recorder.cpp b/services/mediametrics/statsd_recorder.cpp
index 6d5fca0..2e5ada4 100644
--- a/services/mediametrics/statsd_recorder.cpp
+++ b/services/mediametrics/statsd_recorder.cpp
@@ -32,7 +32,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "frameworks/base/core/proto/android/stats/mediametrics/mediametrics.pb.h"
+#include "frameworks/proto_logging/stats/enums/stats/mediametrics/mediametrics.pb.h"
#include "iface_statsd.h"
namespace android {
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index ceefe93..b139be1 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -182,11 +182,12 @@
: AUDIO_SOURCE_DEFAULT;
audio_flags_mask_t flags;
if (direction == AAUDIO_DIRECTION_OUTPUT) {
- flags = AUDIO_FLAG_LOW_LATENCY
- | AAudioConvert_allowCapturePolicyToAudioFlagsMask(params->getAllowedCapturePolicy());
+ flags = static_cast<audio_flags_mask_t>(AUDIO_FLAG_LOW_LATENCY
+ | AAudioConvert_allowCapturePolicyToAudioFlagsMask(
+ params->getAllowedCapturePolicy()));
} else {
- flags = AUDIO_FLAG_LOW_LATENCY
- | AAudioConvert_privacySensitiveToAudioFlagsMask(params->isPrivacySensitive());
+ flags = static_cast<audio_flags_mask_t>(AUDIO_FLAG_LOW_LATENCY
+ | AAudioConvert_privacySensitiveToAudioFlagsMask(params->isPrivacySensitive()));
}
return {
.content_type = contentType,