Merge "Reapply "Add media.c2 AIDL service to media.swcodec"" into main
diff --git a/media/audioaidlconversion/AidlConversionCppNdk.cpp b/media/audioaidlconversion/AidlConversionCppNdk.cpp
index 5558259..947352f 100644
--- a/media/audioaidlconversion/AidlConversionCppNdk.cpp
+++ b/media/audioaidlconversion/AidlConversionCppNdk.cpp
@@ -565,7 +565,8 @@
AudioDeviceType::IN_DEVICE, AudioDeviceType::OUT_DEVICE);
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_PROXY, AUDIO_DEVICE_OUT_PROXY,
- AudioDeviceType::IN_AFE_PROXY, AudioDeviceType::OUT_AFE_PROXY);
+ AudioDeviceType::IN_AFE_PROXY, AudioDeviceType::OUT_AFE_PROXY,
+ GET_DEVICE_DESC_CONNECTION(VIRTUAL));
append_AudioDeviceDescription(pairs,
AUDIO_DEVICE_IN_USB_HEADSET, AUDIO_DEVICE_OUT_USB_HEADSET,
AudioDeviceType::IN_HEADSET, AudioDeviceType::OUT_HEADSET,
diff --git a/media/codec2/sfplugin/CCodec.cpp b/media/codec2/sfplugin/CCodec.cpp
index 8082dd7..cd3a80f 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -2326,9 +2326,12 @@
void CCodec::onWorkDone(std::list<std::unique_ptr<C2Work>> &workItems) {
if (!workItems.empty()) {
Mutexed<std::list<std::unique_ptr<C2Work>>>::Locked queue(mWorkDoneQueue);
+ bool shouldPost = queue->empty();
queue->splice(queue->end(), workItems);
+ if (shouldPost) {
+ (new AMessage(kWhatWorkDone, this))->post();
+ }
}
- (new AMessage(kWhatWorkDone, this))->post();
}
void CCodec::onInputBufferDone(uint64_t frameIndex, size_t arrayIndex) {
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 073a030..c912d27 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -68,10 +68,9 @@
}
// We double the size of input buffer for ping pong use of record buffer.
- // Assumes audio_is_linear_pcm(format)
- const auto sampleSize = audio_channel_count_from_in_mask(channelMask) *
- audio_bytes_per_sample(format);
- if (sampleSize == 0 || ((*frameCount = (size * 2) / sampleSize) == 0)) {
+ const auto frameSize = audio_bytes_per_frame(
+ audio_channel_count_from_in_mask(channelMask), format);
+ if (frameSize == 0 || ((*frameCount = (size * 2) / frameSize) == 0)) {
ALOGE("%s(): Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
__func__, sampleRate, format, channelMask);
return BAD_VALUE;
@@ -353,12 +352,7 @@
}
mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
-
- if (audio_is_linear_pcm(mFormat)) {
- mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
- } else {
- mFrameSize = sizeof(uint8_t);
- }
+ mFrameSize = audio_bytes_per_frame(mChannelCount, mFormat);
// mFrameCount is initialized in createRecord_l
mReqFrameCount = frameCount;
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index f3539a1..f050a20 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -594,18 +594,13 @@
channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
- if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
- if (audio_has_proportional_frames(format)) {
- mFrameSize = channelCount * audio_bytes_per_sample(format);
- } else {
- mFrameSize = sizeof(uint8_t);
- }
- } else {
- ALOG_ASSERT(audio_has_proportional_frames(format));
- mFrameSize = channelCount * audio_bytes_per_sample(format);
+ if (!(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
// createTrack will return an error if PCM format is not supported by server,
// so no need to check for specific PCM formats here
+ ALOGW_IF(!audio_has_proportional_frames(format), "%s(): no direct flag for format 0x%x",
+ __func__, format);
}
+ mFrameSize = audio_bytes_per_frame(channelCount, format);
// sampling rate must be specified for direct outputs
if (sampleRate == 0 && (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
diff --git a/media/libaudiofoundation/include/media/DeviceDescriptorBase.h b/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
index 501831d..2ab8053 100644
--- a/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
+++ b/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
@@ -79,8 +79,8 @@
bool equals(const sp<DeviceDescriptorBase>& other) const;
- status_t writeToParcelable(media::AudioPortFw* parcelable) const;
- status_t readFromParcelable(const media::AudioPortFw& parcelable);
+ virtual status_t writeToParcelable(media::AudioPortFw* parcelable) const;
+ virtual status_t readFromParcelable(const media::AudioPortFw& parcelable);
protected:
AudioDeviceTypeAddr mDeviceTypeAddr;
diff --git a/media/libaudiohal/impl/DeviceHalAidl.cpp b/media/libaudiohal/impl/DeviceHalAidl.cpp
index e2233c7..5bc25ae 100644
--- a/media/libaudiohal/impl/DeviceHalAidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalAidl.cpp
@@ -28,6 +28,7 @@
#include <media/AidlConversionNdkCpp.h>
#include <media/AidlConversionUtil.h>
#include <mediautils/TimeCheck.h>
+#include <system/audio.h>
#include <Utils.h>
#include <utils/Log.h>
@@ -41,6 +42,7 @@
using aidl::android::media::audio::common::AudioConfig;
using aidl::android::media::audio::common::AudioDevice;
using aidl::android::media::audio::common::AudioDeviceAddress;
+using aidl::android::media::audio::common::AudioDeviceDescription;
using aidl::android::media::audio::common::AudioDeviceType;
using aidl::android::media::audio::common::AudioFormatDescription;
using aidl::android::media::audio::common::AudioFormatType;
@@ -185,23 +187,77 @@
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(mModule->getAudioPorts(&ports)));
ALOGW_IF(ports.empty(), "%s: module %s returned an empty list of audio ports",
__func__, mInstance.c_str());
- std::transform(ports.begin(), ports.end(), std::inserter(mPorts, mPorts.end()),
- [](const auto& p) { return std::make_pair(p.id, p); });
mDefaultInputPortId = mDefaultOutputPortId = -1;
const int defaultDeviceFlag = 1 << AudioPortDeviceExt::FLAG_INDEX_DEFAULT_DEVICE;
- for (const auto& pair : mPorts) {
- const auto& p = pair.second;
- if (p.ext.getTag() == AudioPortExt::Tag::device &&
- (p.ext.get<AudioPortExt::Tag::device>().flags & defaultDeviceFlag) != 0) {
- if (p.flags.getTag() == AudioIoFlags::Tag::input) {
- mDefaultInputPortId = p.id;
- } else if (p.flags.getTag() == AudioIoFlags::Tag::output) {
- mDefaultOutputPortId = p.id;
+ for (auto it = ports.begin(); it != ports.end(); ) {
+ const auto& port = *it;
+ if (port.ext.getTag() != AudioPortExt::Tag::device) {
+ ++it;
+ continue;
+ }
+ const AudioPortDeviceExt& deviceExt = port.ext.get<AudioPortExt::Tag::device>();
+ if ((deviceExt.flags & defaultDeviceFlag) != 0) {
+ if (port.flags.getTag() == AudioIoFlags::Tag::input) {
+ mDefaultInputPortId = port.id;
+ } else if (port.flags.getTag() == AudioIoFlags::Tag::output) {
+ mDefaultOutputPortId = port.id;
}
}
+ // For compatibility with HIDL, hide "template" remote submix ports from ports list.
+ if (const auto& devDesc = deviceExt.device;
+ (devDesc.type.type == AudioDeviceType::IN_SUBMIX ||
+ devDesc.type.type == AudioDeviceType::OUT_SUBMIX) &&
+ devDesc.type.connection == AudioDeviceDescription::CONNECTION_VIRTUAL) {
+ if (devDesc.type.type == AudioDeviceType::IN_SUBMIX) {
+ mRemoteSubmixIn = port;
+ } else {
+ mRemoteSubmixOut = port;
+ }
+ it = ports.erase(it);
+ } else {
+ ++it;
+ }
}
+ if (mRemoteSubmixIn.has_value() != mRemoteSubmixOut.has_value()) {
+ ALOGE("%s: The configuration only has input or output remote submix device, must have both",
+ __func__);
+ mRemoteSubmixIn.reset();
+ mRemoteSubmixOut.reset();
+ }
+ if (mRemoteSubmixIn.has_value()) {
+ AudioPort connectedRSubmixIn = *mRemoteSubmixIn;
+ connectedRSubmixIn.ext.get<AudioPortExt::Tag::device>().device.address =
+ AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS;
+ ALOGD("%s: connecting remote submix input", __func__);
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(mModule->connectExternalDevice(
+ connectedRSubmixIn, &connectedRSubmixIn)));
+ // The template port for the remote submix input couldn't be "default" because it is not
+ // attached. The connected port can now be made default because we never disconnect it.
+ if (mDefaultInputPortId == -1) {
+ mDefaultInputPortId = connectedRSubmixIn.id;
+ }
+ ports.push_back(std::move(connectedRSubmixIn));
+
+ // Remote submix output must not be connected until the framework actually starts
+ // using it, however for legacy compatibility we need to provide an "augmented template"
+ // port with an address and profiles. It is obtained by connecting the output and then
+ // immediately disconnecting it. This is a cheap operation as we don't open any streams.
+ AudioPort tempConnectedRSubmixOut = *mRemoteSubmixOut;
+ tempConnectedRSubmixOut.ext.get<AudioPortExt::Tag::device>().device.address =
+ AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS;
+ ALOGD("%s: temporarily connecting and disconnecting remote submix output", __func__);
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(mModule->connectExternalDevice(
+ tempConnectedRSubmixOut, &tempConnectedRSubmixOut)));
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(mModule->disconnectExternalDevice(
+ tempConnectedRSubmixOut.id)));
+ tempConnectedRSubmixOut.id = mRemoteSubmixOut->id;
+ ports.push_back(std::move(tempConnectedRSubmixOut));
+ }
+
ALOGI("%s: module %s default port ids: input %d, output %d",
__func__, mInstance.c_str(), mDefaultInputPortId, mDefaultOutputPortId);
+ std::transform(ports.begin(), ports.end(), std::inserter(mPorts, mPorts.end()),
+ [](const auto& p) { return std::make_pair(p.id, p); });
RETURN_STATUS_IF_ERROR(updateRoutes());
std::vector<AudioPortConfig> portConfigs;
RETURN_STATUS_IF_ERROR(
@@ -1079,6 +1135,7 @@
}
status_t DeviceHalAidl::setConnectedState(const struct audio_port_v7 *port, bool connected) {
+ ALOGD("%p %s::%s", this, getClassName().c_str(), __func__);
TIME_CHECK();
if (!mModule) return NO_INIT;
if (port == nullptr) {
@@ -1104,20 +1161,43 @@
}
if (connected) {
AudioDevice matchDevice = aidlPort.ext.get<AudioPortExt::device>().device;
- // Reset the device address to find the "template" port.
- matchDevice.address = AudioDeviceAddress::make<AudioDeviceAddress::id>();
- auto portsIt = findPort(matchDevice);
- if (portsIt == mPorts.end()) {
- // Since 'setConnectedState' is called for all modules, it is normal when the device
- // port not found in every one of them.
- return BAD_VALUE;
+ std::optional<AudioPort> templatePort;
+ auto erasePortAfterConnectionIt = mPorts.end();
+ // Connection of remote submix out with address "0" is a special case. Since there is
+ // already an "augmented template" port with this address in mPorts, we need to replace
+ // it with a connected port.
+ // Connection of remote submix outs with any other address is done as usual except that
+ // the template port is in `mRemoteSubmixOut`.
+ if (mRemoteSubmixOut.has_value() &&
+ matchDevice.type.type == AudioDeviceType::OUT_SUBMIX) {
+ if (matchDevice.address == AudioDeviceAddress::make<AudioDeviceAddress::id>(
+ AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS)) {
+ erasePortAfterConnectionIt = findPort(matchDevice);
+ }
+ templatePort = mRemoteSubmixOut;
+ } else if (mRemoteSubmixIn.has_value() &&
+ matchDevice.type.type == AudioDeviceType::IN_SUBMIX) {
+ templatePort = mRemoteSubmixIn;
} else {
- ALOGD("%s: device port for device %s found in the module %s",
- __func__, matchDevice.toString().c_str(), mInstance.c_str());
+ // Reset the device address to find the "template" port.
+ matchDevice.address = AudioDeviceAddress::make<AudioDeviceAddress::id>();
+ }
+ if (!templatePort.has_value()) {
+ auto portsIt = findPort(matchDevice);
+ if (portsIt == mPorts.end()) {
+ // Since 'setConnectedState' is called for all modules, it is normal when the device
+ // port not found in every one of them.
+ return BAD_VALUE;
+ } else {
+ ALOGD("%s: device port for device %s found in the module %s",
+ __func__, matchDevice.toString().c_str(), mInstance.c_str());
+ }
+ templatePort = portsIt->second;
}
resetUnusedPatchesAndPortConfigs();
+
// Use the ID of the "template" port, use all the information from the provided port.
- aidlPort.id = portsIt->first;
+ aidlPort.id = templatePort->id;
AudioPort connectedPort;
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(mModule->connectExternalDevice(
aidlPort, &connectedPort)));
@@ -1127,6 +1207,9 @@
__func__, mInstance.c_str(), connectedPort.toString().c_str(),
it->second.toString().c_str());
mConnectedPorts[connectedPort.id] = false;
+ if (erasePortAfterConnectionIt != mPorts.end()) {
+ mPorts.erase(erasePortAfterConnectionIt);
+ }
} else { // !connected
AudioDevice matchDevice = aidlPort.ext.get<AudioPortExt::device>().device;
auto portsIt = findPort(matchDevice);
@@ -1139,14 +1222,24 @@
__func__, matchDevice.toString().c_str(), mInstance.c_str());
}
resetUnusedPatchesAndPortConfigs();
+
+ // Disconnection of remote submix out with address "0" is a special case. We need to replace
+ // the connected port entry with the "augmented template".
+ const int32_t portId = portsIt->second.id;
+ if (mRemoteSubmixOut.has_value() && matchDevice.type.type == AudioDeviceType::OUT_SUBMIX &&
+ matchDevice.address == AudioDeviceAddress::make<AudioDeviceAddress::id>(
+ AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS)) {
+ mDisconnectedPortReplacement = std::make_pair(portId, *mRemoteSubmixOut);
+ auto& port = mDisconnectedPortReplacement.second;
+ port.ext.get<AudioPortExt::Tag::device>().device = matchDevice;
+ port.profiles = portsIt->second.profiles;
+ }
// Streams are closed by AudioFlinger independently from device disconnections.
// It is possible that the stream has not been closed yet.
- const int32_t portId = portsIt->second.id;
if (!isPortHeldByAStream(portId)) {
RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
mModule->disconnectExternalDevice(portId)));
- mPorts.erase(portsIt);
- mConnectedPorts.erase(portId);
+ eraseConnectedPort(portId);
} else {
ALOGD("%s: since device port ID %d is used by a stream, "
"external device disconnection postponed", __func__, portId);
@@ -1216,6 +1309,17 @@
return OK;
}
+void DeviceHalAidl::eraseConnectedPort(int32_t portId) {
+ mPorts.erase(portId);
+ mConnectedPorts.erase(portId);
+ if (mDisconnectedPortReplacement.first == portId) {
+ const auto& port = mDisconnectedPortReplacement.second;
+ mPorts.insert(std::make_pair(port.id, port));
+ ALOGD("%s: disconnected port replacement: %s", __func__, port.toString().c_str());
+ mDisconnectedPortReplacement = std::pair<int32_t, AudioPort>();
+ }
+}
+
status_t DeviceHalAidl::filterAndRetrieveBtA2dpParameters(
AudioParameter &keys, AudioParameter *result) {
TIME_CHECK();
@@ -1847,8 +1951,7 @@
if (!isPortHeldByAStream(portId)) {
TIME_CHECK();
if (auto status = mModule->disconnectExternalDevice(portId); status.isOk()) {
- mPorts.erase(portId);
- mConnectedPorts.erase(portId);
+ eraseConnectedPort(portId);
ALOGD("%s: executed postponed external device disconnection for port ID %d",
__func__, portId);
}
@@ -1865,6 +1968,30 @@
statusTFromBinderStatus(mModule->getAudioRoutes(&mRoutes)));
ALOGW_IF(mRoutes.empty(), "%s: module %s returned an empty list of audio routes",
__func__, mInstance.c_str());
+ if (mRemoteSubmixIn.has_value()) {
+ // Remove mentions of the template remote submix input from routes.
+ int32_t rSubmixInId = mRemoteSubmixIn->id;
+ // Remove mentions of the template remote submix out only if it is not in mPorts
+ // (that means there is a connected port in mPorts).
+ int32_t rSubmixOutId = mPorts.find(mRemoteSubmixOut->id) == mPorts.end() ?
+ mRemoteSubmixOut->id : -1;
+ for (auto it = mRoutes.begin(); it != mRoutes.end();) {
+ auto& route = *it;
+ if (route.sinkPortId == rSubmixOutId) {
+ it = mRoutes.erase(it);
+ continue;
+ }
+ if (auto sourceIt = std::find(route.sourcePortIds.begin(), route.sourcePortIds.end(),
+ rSubmixInId); sourceIt != route.sourcePortIds.end()) {
+ route.sourcePortIds.erase(sourceIt);
+ if (route.sourcePortIds.empty()) {
+ it = mRoutes.erase(it);
+ continue;
+ }
+ }
+ ++it;
+ }
+ }
mRoutingMatrix.clear();
for (const auto& r : mRoutes) {
for (auto portId : r.sourcePortIds) {
diff --git a/media/libaudiohal/impl/DeviceHalAidl.h b/media/libaudiohal/impl/DeviceHalAidl.h
index e1fe4d1..99e8503 100644
--- a/media/libaudiohal/impl/DeviceHalAidl.h
+++ b/media/libaudiohal/impl/DeviceHalAidl.h
@@ -223,6 +223,7 @@
status_t createOrUpdatePortConfig(
const ::aidl::android::media::audio::common::AudioPortConfig& requestedPortConfig,
PortConfigs::iterator* result, bool *created);
+ void eraseConnectedPort(int32_t portId);
status_t filterAndRetrieveBtA2dpParameters(AudioParameter &keys, AudioParameter *result);
status_t filterAndUpdateBtA2dpParameters(AudioParameter ¶meters);
status_t filterAndUpdateBtHfpParameters(AudioParameter ¶meters);
@@ -313,6 +314,10 @@
std::shared_ptr<::aidl::android::hardware::audio::core::sounddose::ISoundDose>
mSoundDose = nullptr;
Ports mPorts;
+ // Remote submix "template" ports (no address specified, no profiles).
+ // They are excluded from `mPorts` as their presence confuses the framework code.
+ std::optional<::aidl::android::media::audio::common::AudioPort> mRemoteSubmixIn;
+ std::optional<::aidl::android::media::audio::common::AudioPort> mRemoteSubmixOut;
int32_t mDefaultInputPortId = -1;
int32_t mDefaultOutputPortId = -1;
PortConfigs mPortConfigs;
@@ -326,6 +331,8 @@
std::map<void*, Callbacks> mCallbacks GUARDED_BY(mLock);
std::set<audio_port_handle_t> mDeviceDisconnectionNotified;
ConnectedPorts mConnectedPorts;
+ std::pair<int32_t, ::aidl::android::media::audio::common::AudioPort>
+ mDisconnectedPortReplacement;
};
} // namespace android
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index df64676..1a37622 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -17,7 +17,7 @@
#ifndef LVM_FLOAT
typedef float LVM_FLOAT;
#endif
-#define LOG_TAG "Bundle"
+#define LOG_TAG "EffectBundle"
#define ARRAY_SIZE(array) (sizeof(array) / sizeof(array)[0])
//#define LOG_NDEBUG 0
@@ -1191,11 +1191,13 @@
// 0 if the configuration is supported
//----------------------------------------------------------------------------
int VirtualizerIsDeviceSupported(audio_devices_t deviceType) {
+ ALOGV("%s: deviceType:%#x", __func__, deviceType);
switch (deviceType) {
case AUDIO_DEVICE_OUT_WIRED_HEADSET:
case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
case AUDIO_DEVICE_OUT_USB_HEADSET:
+ case AUDIO_DEVICE_OUT_BLE_HEADSET:
// case AUDIO_DEVICE_OUT_USB_DEVICE: // For USB testing of the virtualizer only.
return 0;
default:
@@ -3372,10 +3374,10 @@
if (pContext->EffectType == LVM_BASS_BOOST) {
if ((device == AUDIO_DEVICE_OUT_SPEAKER) ||
(device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) ||
- (device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER)) {
- ALOGV("\tEFFECT_CMD_SET_DEVICE device is invalid for LVM_BASS_BOOST %d",
- *(int32_t*)pCmdData);
- ALOGV("\tEFFECT_CMD_SET_DEVICE temporary disable LVM_BAS_BOOST");
+ device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER ||
+ device == AUDIO_DEVICE_OUT_BLE_SPEAKER) {
+ ALOGV("%s: EFFECT_CMD_SET_DEVICE device %#x is invalid for LVM_BASS_BOOST",
+ __func__, device);
// If a device doesn't support bassboost the effect must be temporarily disabled
// the effect must still report its original state as this can only be changed
diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp
index 1cb4410..3abc35e 100644
--- a/media/libnbaio/NBAIO.cpp
+++ b/media/libnbaio/NBAIO.cpp
@@ -55,8 +55,7 @@
ret.mSampleRate = sampleRate;
ret.mChannelCount = channelCount;
ret.mFormat = format;
- ret.mFrameSize = audio_is_linear_pcm(format) ?
- channelCount * audio_bytes_per_sample(format) : sizeof(uint8_t);
+ ret.mFrameSize = audio_bytes_per_frame(channelCount, format);
return ret;
}
diff --git a/media/libstagefright/httplive/fuzzer/Android.bp b/media/libstagefright/httplive/fuzzer/Android.bp
index dd49714..cb2f4ee 100644
--- a/media/libstagefright/httplive/fuzzer/Android.bp
+++ b/media/libstagefright/httplive/fuzzer/Android.bp
@@ -48,6 +48,7 @@
"libstagefright_httplive_headers",
],
shared_libs: [
+ "libbase",
"libcrypto",
"libstagefright_foundation",
"libhidlbase",
diff --git a/media/libstagefright/tests/VideoRenderQualityTracker_test.cpp b/media/libstagefright/tests/VideoRenderQualityTracker_test.cpp
index 7823922..ee76a67 100644
--- a/media/libstagefright/tests/VideoRenderQualityTracker_test.cpp
+++ b/media/libstagefright/tests/VideoRenderQualityTracker_test.cpp
@@ -261,7 +261,7 @@
Configuration::GetServerConfigurableFlagFn getServerConfigurableFlagFn{
[](const std::string &, const std::string &flag, const std::string &) -> std::string {
if (flag == "render_metrics_enabled") {
- return "false";
+ return "true";
} else if (flag == "render_metrics_are_skipped_frames_dropped") {
return "false";
} else if (flag == "render_metrics_max_expected_content_frame_duration_us") {
@@ -304,7 +304,7 @@
// default - if we are accidentally configuring to the default then we're not necessarily
// testing the parsing.
Configuration d;
- EXPECT_EQ(c.enabled, false);
+ EXPECT_EQ(c.enabled, true);
EXPECT_NE(c.enabled, d.enabled);
EXPECT_EQ(c.areSkippedFramesDropped, false);
EXPECT_NE(c.areSkippedFramesDropped, d.areSkippedFramesDropped);
@@ -357,6 +357,7 @@
TEST_F(VideoRenderQualityTrackerTest, countsReleasedFrames) {
Configuration c;
+ c.enabled = true;
Helper h(16.66, c);
h.drop(10);
h.render({16.66, 16.66, 16.66});
@@ -368,6 +369,7 @@
TEST_F(VideoRenderQualityTrackerTest, countsSkippedFrames) {
Configuration c;
+ c.enabled = true;
Helper h(16.66, c);
h.drop(10); // dropped frames are not counted
h.skip(10); // frames skipped before rendering a frame are not counted
@@ -382,6 +384,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenSkippedFramesAreDropped_countsDroppedFrames) {
Configuration c;
+ c.enabled = true;
c.areSkippedFramesDropped = true;
Helper h(16.66, c);
h.skip(10); // skipped frames at the beginning of playback are not counted
@@ -398,6 +401,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenNotSkippedFramesAreDropped_countsDroppedFrames) {
Configuration c;
+ c.enabled = true;
c.areSkippedFramesDropped = false;
Helper h(16.66, c);
h.skip(10); // skipped frames at the beginning of playback are not counted
@@ -414,6 +418,7 @@
TEST_F(VideoRenderQualityTrackerTest, countsRenderedFrames) {
Configuration c;
+ c.enabled = true;
Helper h(16.66, c);
h.drop(10); // dropped frames are not counted
h.render({16.66, 16.66, 16.66});
@@ -425,6 +430,7 @@
TEST_F(VideoRenderQualityTrackerTest, detectsFrameRate) {
Configuration c;
+ c.enabled = true;
c.frameRateDetectionToleranceUs = 2 * 1000; // 2 ms
Helper h(16.66, c);
h.render({16.6, 16.7, 16.6, 16.7});
@@ -434,6 +440,7 @@
TEST_F(VideoRenderQualityTrackerTest, handlesSeeking) {
Configuration c;
+ c.enabled = true;
c.maxExpectedContentFrameDurationUs = 30;
VideoRenderQualityTracker v(c);
v.onFrameReleased(0, 0);
@@ -472,6 +479,7 @@
TEST_F(VideoRenderQualityTrackerTest, withSkipping_handlesSeeking) {
Configuration c;
+ c.enabled = true;
c.maxExpectedContentFrameDurationUs = 30;
VideoRenderQualityTracker v(c);
v.onFrameReleased(0, 0);
@@ -508,6 +516,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenLowTolerance_doesntDetectFrameRate) {
Configuration c;
+ c.enabled = true;
c.frameRateDetectionToleranceUs = 0;
Helper h(16.66, c);
h.render({16.6, 16.7, 16.6, 16.7});
@@ -517,6 +526,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenFrameRateDestabilizes_detectsFrameRate) {
Configuration c;
+ c.enabled = true;
c.frameRateDetectionToleranceUs = 2 * 1000; // 2 ms
Helper h(16.66, c);
h.render({16.6, 16.7, 16.6, 16.7});
@@ -527,6 +537,7 @@
TEST_F(VideoRenderQualityTrackerTest, detects32Pulldown) {
Configuration c;
+ c.enabled = true;
c.frameRateDetectionToleranceUs = 2 * 1000; // 2 ms
Helper h(41.66, c);
h.render({49.9, 33.2, 50.0, 33.4, 50.1, 33.2});
@@ -536,6 +547,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenBad32Pulldown_doesntDetect32Pulldown) {
Configuration c;
+ c.enabled = true;
c.frameRateDetectionToleranceUs = 2 * 1000; // 2 ms
Helper h(41.66, c);
h.render({50.0, 33.33, 33.33, 50.00, 33.33, 50.00});
@@ -545,6 +557,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenFrameRateChanges_detectsMostRecentFrameRate) {
Configuration c;
+ c.enabled = true;
c.frameRateDetectionToleranceUs = 2 * 1000; // 2 ms
Helper h(16.66, c);
h.render({16.6, 16.7, 16.6, 16.7});
@@ -558,6 +571,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenFrameRateIsUnstable_doesntDetectFrameRate) {
Configuration c;
+ c.enabled = true;
c.frameRateDetectionToleranceUs = 2 * 1000; // 2 ms
Helper h(16.66, c);
h.render({16.66, 30.0, 16.66, 30.0, 16.66});
@@ -567,6 +581,7 @@
TEST_F(VideoRenderQualityTrackerTest, capturesFreezeRate) {
Configuration c;
+ c.enabled = true;
Helper h(20, c);
h.render(3);
EXPECT_EQ(h.getMetrics().freezeRate, 0);
@@ -579,6 +594,7 @@
TEST_F(VideoRenderQualityTrackerTest, capturesFreezeDurationHistogram) {
Configuration c;
+ c.enabled = true;
// +17 because freeze durations include the render time of the previous frame
c.freezeDurationMsHistogramBuckets = {2 * 17 + 17, 3 * 17 + 17, 6 * 17 + 17};
Helper h(17, c);
@@ -612,6 +628,7 @@
TEST_F(VideoRenderQualityTrackerTest, capturesFreezeDistanceHistogram) {
Configuration c;
+ c.enabled = true;
c.freezeDistanceMsHistogramBuckets = {1 * 17, 5 * 17, 6 * 17};
Helper h(17, c);
h.render(1);
@@ -643,6 +660,7 @@
TEST_F(VideoRenderQualityTrackerTest, when60hz_hasNoJudder) {
Configuration c;
+ c.enabled = true;
Helper h(16.66, c); // ~24Hz
h.render({16.66, 16.66, 16.66, 16.66, 16.66, 16.66, 16.66});
EXPECT_LE(h.getMetrics().judderScoreHistogram.getMax(), 0);
@@ -651,6 +669,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenSmallVariance60hz_hasNoJudder) {
Configuration c;
+ c.enabled = true;
Helper h(16.66, c); // ~24Hz
h.render({14, 18, 14, 18, 14, 18, 14, 18});
EXPECT_LE(h.getMetrics().judderScoreHistogram.getMax(), 0);
@@ -659,6 +678,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenBadSmallVariance60Hz_hasJudder) {
Configuration c;
+ c.enabled = true;
Helper h(16.66, c); // ~24Hz
h.render({14, 18, 14, /* no 18 between 14s */ 14, 18, 14, 18});
EXPECT_EQ(h.getMetrics().judderScoreHistogram.getCount(), 1);
@@ -666,6 +686,7 @@
TEST_F(VideoRenderQualityTrackerTest, when30Hz_hasNoJudder) {
Configuration c;
+ c.enabled = true;
Helper h(33.33, c);
h.render({33.33, 33.33, 33.33, 33.33, 33.33, 33.33});
EXPECT_LE(h.getMetrics().judderScoreHistogram.getMax(), 0);
@@ -674,6 +695,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenSmallVariance30Hz_hasNoJudder) {
Configuration c;
+ c.enabled = true;
Helper h(33.33, c);
h.render({29.0, 35.0, 29.0, 35.0, 29.0, 35.0});
EXPECT_LE(h.getMetrics().judderScoreHistogram.getMax(), 0);
@@ -682,6 +704,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenBadSmallVariance30Hz_hasJudder) {
Configuration c;
+ c.enabled = true;
Helper h(33.33, c);
h.render({29.0, 35.0, 29.0, /* no 35 between 29s */ 29.0, 35.0, 29.0, 35.0});
EXPECT_EQ(h.getMetrics().judderScoreHistogram.getCount(), 1);
@@ -689,6 +712,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenBad30HzTo60Hz_hasJudder) {
Configuration c;
+ c.enabled = true;
Helper h(33.33, c);
h.render({33.33, 33.33, 50.0, /* frame stayed 1 vsync too long */ 16.66, 33.33, 33.33});
EXPECT_EQ(h.getMetrics().judderScoreHistogram.getCount(), 2); // note: 2 counts of judder
@@ -696,6 +720,7 @@
TEST_F(VideoRenderQualityTrackerTest, when24HzTo60Hz_hasNoJudder) {
Configuration c;
+ c.enabled = true;
Helper h(41.66, c);
h.render({50.0, 33.33, 50.0, 33.33, 50.0, 33.33});
EXPECT_LE(h.getMetrics().judderScoreHistogram.getMax(), 0);
@@ -704,6 +729,7 @@
TEST_F(VideoRenderQualityTrackerTest, when25HzTo60Hz_hasJudder) {
Configuration c;
+ c.enabled = true;
Helper h(40, c);
h.render({33.33, 33.33, 50.0});
h.render({33.33, 33.33, 50.0});
@@ -716,6 +742,7 @@
TEST_F(VideoRenderQualityTrackerTest, when50HzTo60Hz_hasJudder) {
Configuration c;
+ c.enabled = true;
Helper h(20, c);
h.render({16.66, 16.66, 16.66, 33.33});
h.render({16.66, 16.66, 16.66, 33.33});
@@ -728,6 +755,7 @@
TEST_F(VideoRenderQualityTrackerTest, when30HzTo50Hz_hasJudder) {
Configuration c;
+ c.enabled = true;
Helper h(33.33, c);
h.render({40.0, 40.0, 40.0, 60.0});
h.render({40.0, 40.0, 40.0, 60.0});
@@ -739,6 +767,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenSmallVariancePulldown24HzTo60Hz_hasNoJudder) {
Configuration c;
+ c.enabled = true;
Helper h(41.66, c);
h.render({52.0, 31.33, 52.0, 31.33, 52.0, 31.33});
EXPECT_EQ(h.getMetrics().judderScoreHistogram.getCount(), 0);
@@ -746,6 +775,7 @@
TEST_F(VideoRenderQualityTrackerTest, whenBad24HzTo60Hz_hasJudder) {
Configuration c;
+ c.enabled = true;
Helper h(41.66, c);
h.render({50.0, 33.33, 50.0, 33.33, /* no 50 between 33s */ 33.33, 50.0, 33.33});
EXPECT_EQ(h.getMetrics().judderScoreHistogram.getCount(), 1);
@@ -753,6 +783,7 @@
TEST_F(VideoRenderQualityTrackerTest, capturesJudderScoreHistogram) {
Configuration c;
+ c.enabled = true;
c.judderErrorToleranceUs = 2000;
c.judderScoreHistogramBuckets = {1, 5, 8};
Helper h(16, c);
@@ -767,6 +798,7 @@
TEST_F(VideoRenderQualityTrackerTest, ranksJudderScoresInOrder) {
// Each rendering is ranked from best to worst from a user experience
Configuration c;
+ c.enabled = true;
c.judderErrorToleranceUs = 2000;
c.judderScoreHistogramBuckets = {0, 1000};
int64_t previousScore = 0;
@@ -846,6 +878,7 @@
TEST_F(VideoRenderQualityTrackerTest, capturesFreezeEvents) {
Configuration c;
+ c.enabled = true;
c.freezeEventMax = 5;
c.freezeEventDetailsMax = 4;
c.freezeEventDistanceToleranceMs = 1000;
@@ -938,6 +971,7 @@
TEST_F(VideoRenderQualityTrackerTest, capturesJudderEvents) {
Configuration c;
+ c.enabled = true;
c.judderEventMax = 4;
c.judderEventDetailsMax = 3;
c.judderEventDistanceToleranceMs = 100;
@@ -988,6 +1022,7 @@
TEST_F(VideoRenderQualityTrackerTest, capturesOverallFreezeScore) {
Configuration c;
+ c.enabled = true;
// # drops * 20ms + 20ms because current frame is frozen + 1 for bucket threshold
c.freezeDurationMsHistogramBuckets = {1 * 20 + 21, 5 * 20 + 21, 10 * 20 + 21};
c.freezeDurationMsHistogramToScore = {10, 100, 1000};
@@ -1012,6 +1047,7 @@
TEST_F(VideoRenderQualityTrackerTest, capturesOverallJudderScore) {
Configuration c;
+ c.enabled = true;
c.judderScoreHistogramBuckets = {0, 6, 10};
c.judderScoreHistogramToScore = {10, 100, 1000};
Helper h(20, c);
diff --git a/media/module/mpeg2ts/ATSParser.cpp b/media/module/mpeg2ts/ATSParser.cpp
index 6aeea3b..86187bd 100644
--- a/media/module/mpeg2ts/ATSParser.cpp
+++ b/media/module/mpeg2ts/ATSParser.cpp
@@ -23,8 +23,8 @@
#include "ESQueue.h"
#include <android/hardware/cas/native/1.0/IDescrambler.h>
-#include <android/hidl/allocator/1.0/IAllocator.h>
#include <android/hidl/memory/1.0/IMemory.h>
+#include <cutils/ashmem.h>
#include <cutils/native_handle.h>
#include <hidlmemory/mapping.h>
#include <media/cas/DescramblerAPI.h>
@@ -46,12 +46,12 @@
#include <inttypes.h>
namespace android {
+using hardware::hidl_handle;
using hardware::hidl_string;
using hardware::hidl_vec;
using hardware::hidl_memory;
using namespace hardware::cas::V1_0;
using namespace hardware::cas::native::V1_0;
-typedef hidl::allocator::V1_0::IAllocator TAllocator;
typedef hidl::memory::V1_0::IMemory TMemory;
// I want the expression "y" evaluated even if verbose logging is off.
@@ -210,7 +210,6 @@
bool mSampleEncrypted;
sp<AMessage> mSampleAesKeyItem;
sp<TMemory> mHidlMemory;
- sp<TAllocator> mHidlAllocator;
hardware::cas::native::V1_0::SharedBuffer mDescramblerSrcBuffer;
sp<ABuffer> mDescrambledBuffer;
List<SubSampleInfo> mSubSamples;
@@ -1006,34 +1005,29 @@
sp<ABuffer> newBuffer, newScrambledBuffer;
sp<TMemory> newMem;
if (mScrambled) {
- if (mHidlAllocator == nullptr) {
- mHidlAllocator = TAllocator::getService("ashmem");
- if (mHidlAllocator == nullptr) {
- ALOGE("[stream %d] can't get hidl allocator", mElementaryPID);
- return false;
+ int fd = ashmem_create_region("mediaATS", neededSize);
+ if (fd < 0) {
+ ALOGE("[stream %d] create_ashmem_region failed for size %zu. FD returned: %d",
+ mElementaryPID, neededSize, fd);
+ return false;
+ }
+
+ native_handle_t* handle = native_handle_create(1 /*numFds*/, 0/*numInts*/);
+ if (handle == nullptr) {
+ ALOGE("[stream %d] failed to create a native_handle_t", mElementaryPID);
+ if (close(fd)) {
+ ALOGE("[stream %d] failed to close ashmem fd. errno: %s", mElementaryPID,
+ strerror(errno));
}
- }
- hidl_memory hidlMemToken;
- bool success;
- auto transStatus = mHidlAllocator->allocate(
- neededSize,
- [&success, &hidlMemToken](
- bool s,
- hidl_memory const& m) {
- success = s;
- hidlMemToken = m;
- });
-
- if (!transStatus.isOk()) {
- ALOGE("[stream %d] hidl allocator failed at the transport: %s",
- mElementaryPID, transStatus.description().c_str());
return false;
}
- if (!success) {
- ALOGE("[stream %d] hidl allocator failed", mElementaryPID);
- return false;
- }
+
+ handle->data[0] = fd;
+ hidl_handle memHandle;
+ memHandle.setTo(handle, true /*shouldOwn*/);
+ hidl_memory hidlMemToken("ashmem", memHandle, neededSize);
+
newMem = mapMemory(hidlMemToken);
if (newMem == nullptr || newMem->getPointer() == nullptr) {
ALOGE("[stream %d] hidl failed to map memory", mElementaryPID);
diff --git a/media/module/mpeg2ts/Android.bp b/media/module/mpeg2ts/Android.bp
index bf762c6..c710ffb 100644
--- a/media/module/mpeg2ts/Android.bp
+++ b/media/module/mpeg2ts/Android.bp
@@ -44,7 +44,6 @@
"libhidlmemory",
"android.hardware.cas.native@1.0",
"android.hidl.memory@1.0",
- "android.hidl.allocator@1.0",
],
header_libs: [
diff --git a/media/module/mpeg2ts/test/Android.bp b/media/module/mpeg2ts/test/Android.bp
index 34a8d3e..cccefac 100644
--- a/media/module/mpeg2ts/test/Android.bp
+++ b/media/module/mpeg2ts/test/Android.bp
@@ -37,9 +37,8 @@
shared_libs: [
"android.hardware.cas@1.0",
"android.hardware.cas.native@1.0",
- "android.hidl.token@1.0-utils",
- "android.hidl.allocator@1.0",
"libcrypto",
+ "libcutils",
"libhidlbase",
"libhidlmemory",
"liblog",
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 29c6845..46384ef 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -27,6 +27,7 @@
//#define BUFLOG_NDEBUG 0
#include <afutils/BufLog.h>
#include <afutils/DumpTryLock.h>
+#include <afutils/NBAIO_Tee.h>
#include <afutils/Permission.h>
#include <afutils/PropertyUtils.h>
#include <afutils/TypedLogger.h>
@@ -2090,7 +2091,8 @@
}
}
-void AudioFlinger::ioConfigChanged(audio_io_config_event_t event,
+// Hold either AudioFlinger::mutex or ThreadBase::mutex
+void AudioFlinger::ioConfigChanged_l(audio_io_config_event_t event,
const sp<AudioIoDescriptor>& ioDesc,
pid_t pid) {
media::AudioIoConfigEvent eventAidl = VALUE_OR_FATAL(
@@ -2944,7 +2946,7 @@
latencyMs = playbackThread->latency();
// notify client processes of the new output creation
- playbackThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
+ playbackThread->ioConfigChanged_l(AUDIO_OUTPUT_OPENED);
// the first primary output opened designates the primary hw device if no HW module
// named "primary" was already loaded.
@@ -2958,7 +2960,7 @@
mHardwareStatus = AUDIO_HW_IDLE;
}
} else {
- thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
+ thread->ioConfigChanged_l(AUDIO_OUTPUT_OPENED);
}
response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
response->config = VALUE_OR_RETURN_STATUS(
@@ -2991,7 +2993,7 @@
thread->addOutputTrack(thread2);
mPlaybackThreads.add(id, thread);
// notify client processes of the new output creation
- thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
+ thread->ioConfigChanged_l(AUDIO_OUTPUT_OPENED);
return id;
}
@@ -3051,7 +3053,7 @@
mMmapThreads.removeItem(output);
ALOGD("closing mmapThread %p", mmapThread.get());
}
- ioConfigChanged(AUDIO_OUTPUT_CLOSED, sp<AudioIoDescriptor>::make(output));
+ ioConfigChanged_l(AUDIO_OUTPUT_CLOSED, sp<AudioIoDescriptor>::make(output));
mPatchPanel->notifyStreamClosed(output);
}
// The thread entity (active unit of execution) is no longer running here,
@@ -3154,7 +3156,7 @@
if (thread != 0) {
// notify client processes of the new input creation
- thread->ioConfigChanged(AUDIO_INPUT_OPENED);
+ thread->ioConfigChanged_l(AUDIO_INPUT_OPENED);
return NO_ERROR;
}
return NO_INIT;
@@ -3313,7 +3315,7 @@
dumpToThreadLog_l(mmapThread);
mMmapThreads.removeItem(input);
}
- ioConfigChanged(AUDIO_INPUT_CLOSED, sp<AudioIoDescriptor>::make(input));
+ ioConfigChanged_l(AUDIO_INPUT_CLOSED, sp<AudioIoDescriptor>::make(input));
}
// FIXME: calling thread->exit() without mutex() held should not be needed anymore now that
// we have a different lock for notification client
@@ -3693,7 +3695,8 @@
return {};
}
- return thread->outDeviceTypes();
+ audio_utils::lock_guard l(thread->mutex());
+ return thread->outDeviceTypes_l();
}
IAfPlaybackThread* AudioFlinger::fastPlaybackThread_l() const
@@ -4487,8 +4490,9 @@
}
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
- sp<IAfEffectChain> ec =
- mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
+ const auto thread = mPlaybackThreads.valueAt(i);
+ audio_utils::lock_guard l(thread->mutex());
+ const sp<IAfEffectChain> ec = thread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
if (ec != 0 && ec->isNonOffloadableEnabled()) {
return true;
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index e7f9255..b1751da 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -342,7 +342,8 @@
// ----- begin IAfThreadCallback interface
- bool isNonOffloadableGlobalEffectEnabled_l() const final REQUIRES(mutex());
+ bool isNonOffloadableGlobalEffectEnabled_l() const final
+ REQUIRES(mutex()) EXCLUDES_ThreadBase_Mutex;
bool btNrecIsOff() const final { return mBtNrecIsOff.load(); }
float masterVolume_l() const final REQUIRES(mutex());
bool masterMute_l() const final REQUIRES(mutex());
@@ -387,7 +388,8 @@
const audioflinger::SyncEventCallback& callBack,
const wp<IAfTrackBase>& cookie) final EXCLUDES_AudioFlinger_Mutex;
- void ioConfigChanged(audio_io_config_event_t event,
+ // Hold either AudioFlinger::mutex or ThreadBase::mutex
+ void ioConfigChanged_l(audio_io_config_event_t event,
const sp<AudioIoDescriptor>& ioDesc,
pid_t pid = 0) final EXCLUDES_AudioFlinger_ClientMutex;
void onNonOffloadableGlobalEffectEnable() final EXCLUDES_AudioFlinger_Mutex;
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 0f3e130..95fed5b 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -2570,6 +2570,7 @@
uint32_t rightZero = 0;
volumeControlEffect->setVolume(&leftZero, &rightZero, true /*controller*/);
}
+ mVolumeControlEffect = volumeControlEffect;
}
mLeftVolume = newLeft;
mRightVolume = newRight;
@@ -3084,7 +3085,10 @@
return t->sampleRate();
}
-audio_channel_mask_t EffectChain::EffectCallback::inChannelMask(int id) const {
+audio_channel_mask_t EffectChain::EffectCallback::inChannelMask(int id) const
+NO_THREAD_SAFETY_ANALYSIS
+// calling function 'hasAudioSession_l' requires holding mutex 'ThreadBase_Mutex' exclusively
+{
const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return AUDIO_CHANNEL_NONE;
@@ -3120,7 +3124,10 @@
return audio_channel_count_from_out_mask(inChannelMask(id));
}
-audio_channel_mask_t EffectChain::EffectCallback::outChannelMask() const {
+audio_channel_mask_t EffectChain::EffectCallback::outChannelMask() const
+NO_THREAD_SAFETY_ANALYSIS
+// calling function 'hasAudioSession_l' requires holding mutex 'ThreadBase_Mutex' exclusively
+{
const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return AUDIO_CHANNEL_NONE;
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index aeb0fea..8869b69 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -488,8 +488,10 @@
bool isBitPerfectCompatible() const final;
// isCompatibleWithThread_l() must be called with thread->mutex() held
- bool isCompatibleWithThread_l(const sp<IAfThreadBase>& thread) const final;
+ bool isCompatibleWithThread_l(const sp<IAfThreadBase>& thread) const final
+ REQUIRES(audio_utils::ThreadBase_Mutex);
+ // Requires either IAfThreadBase::mutex() or EffectChain::mutex() held
bool containsHapticGeneratingEffect_l() final;
void setHapticIntensity_l(int id, os::HapticScale intensity) final;
@@ -648,7 +650,7 @@
const sp<EffectCallback> mEffectCallback;
- wp<EffectModule> mVolumeControlEffect;
+ wp<IAfEffectModule> mVolumeControlEffect;
};
class DeviceEffectProxy : public IAfDeviceEffectProxy, public EffectBase {
diff --git a/services/audioflinger/IAfThread.h b/services/audioflinger/IAfThread.h
index fc2f805..8b41740 100644
--- a/services/audioflinger/IAfThread.h
+++ b/services/audioflinger/IAfThread.h
@@ -70,7 +70,7 @@
virtual audio_utils::mutex& mutex() const
RETURN_CAPABILITY(audio_utils::AudioFlinger_Mutex) = 0;
virtual bool isNonOffloadableGlobalEffectEnabled_l() const
- REQUIRES(mutex()) = 0; // Tracks
+ REQUIRES(mutex()) EXCLUDES_ThreadBase_Mutex = 0; // Tracks
virtual audio_unique_id_t nextUniqueId(audio_unique_id_use_t use) = 0;
virtual bool btNrecIsOff() const = 0;
virtual float masterVolume_l() const
@@ -110,7 +110,8 @@
const wp<IAfTrackBase>& cookie)
EXCLUDES_AudioFlinger_Mutex = 0;
- virtual void ioConfigChanged(audio_io_config_event_t event,
+ // Hold either AudioFlinger::mutex or ThreadBase::mutex
+ virtual void ioConfigChanged_l(audio_io_config_event_t event,
const sp<AudioIoDescriptor>& ioDesc,
pid_t pid = 0) EXCLUDES_AudioFlinger_ClientMutex = 0;
virtual void onNonOffloadableGlobalEffectEnable() EXCLUDES_AudioFlinger_Mutex = 0;
@@ -140,7 +141,7 @@
static bool isValidPcmSinkFormat(audio_format_t format);
virtual status_t readyToRun() = 0;
- virtual void clearPowerManager() = 0;
+ virtual void clearPowerManager() EXCLUDES_ThreadBase_Mutex = 0;
virtual status_t initCheck() const = 0;
virtual type_t type() const = 0;
virtual bool isDuplicating() const = 0;
@@ -156,21 +157,23 @@
virtual size_t frameCount() const = 0;
virtual audio_channel_mask_t hapticChannelMask() const = 0;
virtual uint32_t hapticChannelCount() const = 0;
- virtual uint32_t latency_l() const = 0;
- virtual void setVolumeForOutput_l(float left, float right) const = 0;
+ virtual uint32_t latency_l() const = 0; // NO_THREAD_SAFETY_ANALYSIS
+ virtual void setVolumeForOutput_l(float left, float right) const REQUIRES(mutex()) = 0;
// Return's the HAL's frame count i.e. fast mixer buffer size.
virtual size_t frameCountHAL() const = 0;
virtual size_t frameSize() const = 0;
// Should be "virtual status_t requestExitAndWait()" and override same
// method in Thread, but Thread::requestExitAndWait() is not yet virtual.
- virtual void exit() = 0;
- virtual bool checkForNewParameter_l(const String8& keyValuePair, status_t& status) = 0;
- virtual status_t setParameters(const String8& keyValuePairs) = 0;
- virtual String8 getParameters(const String8& keys) = 0;
- virtual void ioConfigChanged(
+ virtual void exit() EXCLUDES_ThreadBase_Mutex = 0;
+ virtual bool checkForNewParameter_l(const String8& keyValuePair, status_t& status)
+ REQUIRES(mutex()) = 0;
+ virtual status_t setParameters(const String8& keyValuePairs) EXCLUDES_ThreadBase_Mutex = 0;
+ virtual String8 getParameters(const String8& keys) EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void ioConfigChanged_l(
audio_io_config_event_t event, pid_t pid = 0,
- audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0;
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE)
+ /* holds either AF::mutex or TB::mutex */ = 0;
// sendConfigEvent_l() must be called with ThreadBase::mLock held
// Can temporarily release the lock if waiting for a reply from
@@ -178,38 +181,53 @@
// status_t sendConfigEvent_l(sp<ConfigEvent>& event);
virtual void sendIoConfigEvent(
audio_io_config_event_t event, pid_t pid = 0,
- audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0;
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) EXCLUDES_ThreadBase_Mutex = 0;
virtual void sendIoConfigEvent_l(
audio_io_config_event_t event, pid_t pid = 0,
- audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0;
- virtual void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) = 0;
- virtual void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp) = 0;
- virtual status_t sendSetParameterConfigEvent_l(const String8& keyValuePair) = 0;
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) REQUIRES(mutex()) = 0;
+ virtual void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp)
+ EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp)
+ REQUIRES(mutex()) = 0;
+ virtual status_t sendSetParameterConfigEvent_l(const String8& keyValuePair)
+ REQUIRES(mutex()) = 0;
virtual status_t sendCreateAudioPatchConfigEvent(
- const struct audio_patch* patch, audio_patch_handle_t* handle) = 0;
- virtual status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle) = 0;
+ const struct audio_patch* patch, audio_patch_handle_t* handle)
+ EXCLUDES_ThreadBase_Mutex = 0;
+ virtual status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle)
+ EXCLUDES_ThreadBase_Mutex = 0;
virtual status_t sendUpdateOutDeviceConfigEvent(
- const DeviceDescriptorBaseVector& outDevices) = 0;
- virtual void sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs) = 0;
- virtual void sendCheckOutputStageEffectsEvent() = 0;
- virtual void sendCheckOutputStageEffectsEvent_l() = 0;
- virtual void sendHalLatencyModesChangedEvent_l() = 0;
+ const DeviceDescriptorBaseVector& outDevices) EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs)
+ REQUIRES(mutex()) = 0;
+ virtual void sendCheckOutputStageEffectsEvent() EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void sendCheckOutputStageEffectsEvent_l()
+ REQUIRES(mutex()) = 0;
+ virtual void sendHalLatencyModesChangedEvent_l()
+ REQUIRES(mutex()) = 0;
- virtual void processConfigEvents_l() = 0;
- virtual void setCheckOutputStageEffects() = 0;
- virtual void cacheParameters_l() = 0;
+ virtual void processConfigEvents_l()
+ REQUIRES(mutex()) = 0;
+ virtual void setCheckOutputStageEffects() = 0; // no mutex needed
+ virtual void cacheParameters_l()
+ REQUIRES(mutex()) = 0;
virtual status_t createAudioPatch_l(
- const struct audio_patch* patch, audio_patch_handle_t* handle) = 0;
- virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0;
- virtual void updateOutDevices(const DeviceDescriptorBaseVector& outDevices) = 0;
- virtual void toAudioPortConfig(struct audio_port_config* config) = 0;
- virtual void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) = 0;
+ const struct audio_patch* patch, audio_patch_handle_t* handle)
+ REQUIRES(mutex()) = 0;
+ virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle)
+ REQUIRES(mutex()) = 0;
+ virtual void updateOutDevices(const DeviceDescriptorBaseVector& outDevices)
+ EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void toAudioPortConfig(struct audio_port_config* config)
+ EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs)
+ REQUIRES(mutex()) = 0;
// see note at declaration of mStandby, mOutDevice and mInDevice
virtual bool inStandby() const = 0;
- virtual const DeviceTypeSet outDeviceTypes() const = 0;
- virtual audio_devices_t inDeviceType() const = 0;
- virtual DeviceTypeSet getDeviceTypes() const = 0;
+ virtual const DeviceTypeSet outDeviceTypes_l() const REQUIRES(mutex()) = 0;
+ virtual audio_devices_t inDeviceType_l() const REQUIRES(mutex()) = 0;
+ virtual DeviceTypeSet getDeviceTypes_l() const REQUIRES(mutex()) = 0;
virtual const AudioDeviceTypeAddrVector& outDeviceTypeAddrs() const = 0;
virtual const AudioDeviceTypeAddr& inDeviceTypeAddr() const = 0;
virtual bool isOutput() const = 0;
@@ -226,7 +244,7 @@
bool pinned,
bool probe,
bool notifyFramesProcessed)
- REQUIRES(audio_utils::AudioFlinger_Mutex) = 0;
+ REQUIRES(audio_utils::AudioFlinger_Mutex) EXCLUDES_ThreadBase_Mutex = 0;
// return values for hasAudioSession (bit field)
enum effect_state {
@@ -243,28 +261,39 @@
};
// get effect chain corresponding to session Id.
- virtual sp<IAfEffectChain> getEffectChain(audio_session_t sessionId) const = 0;
+ virtual sp<IAfEffectChain> getEffectChain(audio_session_t sessionId) const
+ EXCLUDES_ThreadBase_Mutex = 0;
// same as getEffectChain() but must be called with ThreadBase mutex locked
- virtual sp<IAfEffectChain> getEffectChain_l(audio_session_t sessionId) const = 0;
- virtual std::vector<int> getEffectIds_l(audio_session_t sessionId) const = 0;
+ virtual sp<IAfEffectChain> getEffectChain_l(audio_session_t sessionId) const
+ REQUIRES(mutex()) = 0;
+ virtual std::vector<int> getEffectIds_l(audio_session_t sessionId) const
+ REQUIRES(mutex()) = 0;
// add an effect chain to the chain list (mEffectChains)
- virtual status_t addEffectChain_l(const sp<IAfEffectChain>& chain) = 0;
+ virtual status_t addEffectChain_l(const sp<IAfEffectChain>& chain)
+ REQUIRES(mutex()) = 0;
// remove an effect chain from the chain list (mEffectChains)
- virtual size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) = 0;
+ virtual size_t removeEffectChain_l(const sp<IAfEffectChain>& chain)
+ REQUIRES(mutex()) = 0;
// lock all effect chains Mutexes. Must be called before releasing the
// ThreadBase mutex before processing the mixer and effects. This guarantees the
// integrity of the chains during the process.
// Also sets the parameter 'effectChains' to current value of mEffectChains.
- virtual void lockEffectChains_l(Vector<sp<IAfEffectChain>>& effectChains) = 0;
+ virtual void lockEffectChains_l(Vector<sp<IAfEffectChain>>& effectChains)
+ REQUIRES(mutex()) = 0;
// unlock effect chains after process
- virtual void unlockEffectChains(const Vector<sp<IAfEffectChain>>& effectChains) = 0;
+ virtual void unlockEffectChains(const Vector<sp<IAfEffectChain>>& effectChains)
+ EXCLUDES_ThreadBase_Mutex = 0;
// get a copy of mEffectChains vector
- virtual Vector<sp<IAfEffectChain>> getEffectChains_l() const = 0;
+ virtual Vector<sp<IAfEffectChain>> getEffectChains_l() const
+ REQUIRES(mutex()) = 0;
// set audio mode to all effect chains
- virtual void setMode(audio_mode_t mode) = 0;
+ virtual void setMode(audio_mode_t mode)
+ EXCLUDES_ThreadBase_Mutex = 0;
// get effect module with corresponding ID on specified audio session
- virtual sp<IAfEffectModule> getEffect(audio_session_t sessionId, int effectId) const = 0;
- virtual sp<IAfEffectModule> getEffect_l(audio_session_t sessionId, int effectId) const = 0;
+ virtual sp<IAfEffectModule> getEffect(audio_session_t sessionId, int effectId) const
+ EXCLUDES_ThreadBase_Mutex = 0;
+ virtual sp<IAfEffectModule> getEffect_l(audio_session_t sessionId, int effectId) const
+ REQUIRES(mutex()) = 0;
// add and effect module. Also creates the effect chain is none exists for
// the effects audio session. Only called in a context of moving an effect
// from one thread to another
@@ -272,29 +301,36 @@
REQUIRES(audio_utils::AudioFlinger_Mutex, mutex()) = 0;
// remove and effect module. Also removes the effect chain is this was the last
// effect
- virtual void removeEffect_l(const sp<IAfEffectModule>& effect, bool release = false) = 0;
+ virtual void removeEffect_l(const sp<IAfEffectModule>& effect, bool release = false)
+ REQUIRES(mutex()) = 0;
// disconnect an effect handle from module and destroy module if last handle
- virtual void disconnectEffectHandle(IAfEffectHandle* handle, bool unpinIfLast) = 0;
+ virtual void disconnectEffectHandle(IAfEffectHandle* handle, bool unpinIfLast)
+ EXCLUDES_ThreadBase_Mutex = 0;
// detach all tracks connected to an auxiliary effect
- virtual void detachAuxEffect_l(int effectId) = 0;
+ virtual void detachAuxEffect_l(int effectId) REQUIRES(mutex()) = 0;
// returns a combination of:
// - EFFECT_SESSION if effects on this audio session exist in one chain
// - TRACK_SESSION if tracks on this audio session exist
// - FAST_SESSION if fast tracks on this audio session exist
// - SPATIALIZED_SESSION if spatialized tracks on this audio session exist
- virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0;
- virtual uint32_t hasAudioSession(audio_session_t sessionId) const = 0;
+ virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const REQUIRES(mutex()) = 0;
+ virtual uint32_t hasAudioSession(audio_session_t sessionId) const
+ EXCLUDES_ThreadBase_Mutex = 0;
// the value returned by default implementation is not important as the
// strategy is only meaningful for PlaybackThread which implements this method
- virtual product_strategy_t getStrategyForSession_l(audio_session_t sessionId) const = 0;
+ virtual product_strategy_t getStrategyForSession_l(audio_session_t sessionId) const
+ REQUIRES(mutex()) = 0;
// check if some effects must be suspended/restored when an effect is enabled
// or disabled
virtual void checkSuspendOnEffectEnabled(
- bool enabled, audio_session_t sessionId, bool threadLocked) = 0;
+ bool enabled, audio_session_t sessionId, bool threadLocked)
+ EXCLUDES_ThreadBase_Mutex = 0;
- virtual status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) = 0;
+ virtual status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event)
+ EXCLUDES_ThreadBase_Mutex = 0;
+ // internally static, perhaps make static member.
virtual bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const = 0;
// Return a reference to a per-thread heap which can be used to allocate IMemory
@@ -307,33 +343,35 @@
virtual sp<IMemory> pipeMemory() const = 0;
- virtual void systemReady() = 0;
+ virtual void systemReady() EXCLUDES_ThreadBase_Mutex = 0;
// checkEffectCompatibility_l() must be called with ThreadBase::mLock held
virtual status_t checkEffectCompatibility_l(
- const effect_descriptor_t* desc, audio_session_t sessionId) = 0;
+ const effect_descriptor_t* desc, audio_session_t sessionId) REQUIRES(mutex()) = 0;
- virtual void broadcast_l() = 0;
+ virtual void broadcast_l() REQUIRES(mutex()) = 0;
- virtual bool isTimestampCorrectionEnabled() const = 0;
+ virtual bool isTimestampCorrectionEnabled_l() const REQUIRES(mutex()) = 0;
virtual bool isMsdDevice() const = 0;
- virtual void dump(int fd, const Vector<String16>& args) = 0;
+ virtual void dump(int fd, const Vector<String16>& args) EXCLUDES_ThreadBase_Mutex = 0;
// deliver stats to mediametrics.
- virtual void sendStatistics(bool force) = 0;
+ virtual void sendStatistics(bool force) EXCLUDES_ThreadBase_Mutex = 0;
virtual audio_utils::mutex& mutex() const
RETURN_CAPABILITY(audio_utils::ThreadBase_Mutex) = 0;
- virtual void onEffectEnable(const sp<IAfEffectModule>& effect) = 0;
- virtual void onEffectDisable() = 0;
+ virtual void onEffectEnable(const sp<IAfEffectModule>& effect) EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void onEffectDisable() EXCLUDES_ThreadBase_Mutex = 0;
// invalidateTracksForAudioSession_l must be called with holding mLock.
- virtual void invalidateTracksForAudioSession_l(audio_session_t sessionId) const = 0;
+ virtual void invalidateTracksForAudioSession_l(audio_session_t sessionId) const
+ REQUIRES(mutex()) = 0;
// Invalidate all the tracks with the given audio session.
- virtual void invalidateTracksForAudioSession(audio_session_t sessionId) const = 0;
+ virtual void invalidateTracksForAudioSession(audio_session_t sessionId) const
+ EXCLUDES_ThreadBase_Mutex = 0;
virtual bool isStreamInitialized() const = 0;
virtual void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor)
@@ -341,10 +379,12 @@
virtual void stopMelComputation_l()
REQUIRES(audio_utils::AudioFlinger_Mutex) = 0;
- virtual product_strategy_t getStrategyForStream(audio_stream_type_t stream) const = 0;
+ virtual product_strategy_t getStrategyForStream(audio_stream_type_t stream) const
+ EXCLUDES_AUDIO_ALL = 0;
virtual void setEffectSuspended_l(
- const effect_uuid_t* type, bool suspend, audio_session_t sessionId) = 0;
+ const effect_uuid_t* type, bool suspend, audio_session_t sessionId)
+ REQUIRES(mutex()) = 0;
// Dynamic cast to derived interface
virtual sp<IAfDirectOutputThread> asIAfDirectOutputThread() { return nullptr; }
@@ -392,7 +432,7 @@
// return estimated latency in milliseconds, as reported by HAL
virtual uint32_t latency() const = 0; // should be in IAfThreadBase?
- virtual uint32_t& fastTrackAvailMask_l() = 0;
+ virtual uint32_t& fastTrackAvailMask_l() REQUIRES(mutex()) = 0;
virtual sp<IAfTrack> createTrack_l(
const sp<Client>& client,
@@ -418,12 +458,12 @@
bool isBitPerfect)
REQUIRES(audio_utils::AudioFlinger_Mutex) = 0;
- virtual status_t addTrack_l(const sp<IAfTrack>& track) = 0;
- virtual bool destroyTrack_l(const sp<IAfTrack>& track) = 0;
- virtual bool isTrackActive(const sp<IAfTrack>& track) const = 0;
- virtual void addOutputTrack_l(const sp<IAfTrack>& track) = 0;
+ virtual status_t addTrack_l(const sp<IAfTrack>& track) REQUIRES(mutex()) = 0;
+ virtual bool destroyTrack_l(const sp<IAfTrack>& track) REQUIRES(mutex()) = 0;
+ virtual bool isTrackActive(const sp<IAfTrack>& track) const REQUIRES(mutex()) = 0;
+ virtual void addOutputTrack_l(const sp<IAfTrack>& track) REQUIRES(mutex()) = 0;
- virtual AudioStreamOut* getOutput_l() const = 0;
+ virtual AudioStreamOut* getOutput_l() const REQUIRES(mutex()) = 0;
virtual AudioStreamOut* getOutput() const = 0;
virtual AudioStreamOut* clearOutput() = 0;
@@ -431,50 +471,57 @@
virtual void suspend() = 0;
virtual void restore() = 0;
virtual bool isSuspended() const = 0;
- virtual status_t getRenderPosition(uint32_t* halFrames, uint32_t* dspFrames) const = 0;
+ virtual status_t getRenderPosition(uint32_t* halFrames, uint32_t* dspFrames) const
+ EXCLUDES_ThreadBase_Mutex = 0;
// Consider also removing and passing an explicit mMainBuffer initialization
// parameter to AF::IAfTrack::Track().
virtual float* sinkBuffer() const = 0;
- virtual status_t attachAuxEffect(const sp<IAfTrack>& track, int EffectId) = 0;
- virtual status_t attachAuxEffect_l(const sp<IAfTrack>& track, int EffectId) = 0;
+ virtual status_t attachAuxEffect(const sp<IAfTrack>& track, int EffectId)
+ EXCLUDES_ThreadBase_Mutex = 0;
+ virtual status_t attachAuxEffect_l(const sp<IAfTrack>& track, int EffectId)
+ REQUIRES(mutex()) = 0;
// called with AudioFlinger lock held
- virtual bool invalidateTracks_l(audio_stream_type_t streamType) = 0;
- virtual bool invalidateTracks_l(std::set<audio_port_handle_t>& portIds) = 0;
- virtual void invalidateTracks(audio_stream_type_t streamType) = 0;
+ virtual bool invalidateTracks_l(audio_stream_type_t streamType) REQUIRES(mutex()) = 0;
+ virtual bool invalidateTracks_l(std::set<audio_port_handle_t>& portIds) REQUIRES(mutex()) = 0;
+ virtual void invalidateTracks(audio_stream_type_t streamType)
+ EXCLUDES_ThreadBase_Mutex = 0;
// Invalidate tracks by a set of port ids. The port id will be removed from
// the given set if the corresponding track is found and invalidated.
- virtual void invalidateTracks(std::set<audio_port_handle_t>& portIds) = 0;
+ virtual void invalidateTracks(std::set<audio_port_handle_t>& portIds)
+ EXCLUDES_ThreadBase_Mutex = 0;
- virtual status_t getTimestamp_l(AudioTimestamp& timestamp) = 0;
- virtual void addPatchTrack(const sp<IAfPatchTrack>& track) = 0;
- virtual void deletePatchTrack(const sp<IAfPatchTrack>& track) = 0;
+ virtual status_t getTimestamp_l(AudioTimestamp& timestamp) REQUIRES(mutex()) = 0;
+ virtual void addPatchTrack(const sp<IAfPatchTrack>& track) EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void deletePatchTrack(const sp<IAfPatchTrack>& track) EXCLUDES_ThreadBase_Mutex = 0;
// Return the asynchronous signal wait time.
- virtual int64_t computeWaitTimeNs_l() const = 0;
+ virtual int64_t computeWaitTimeNs_l() const REQUIRES(mutex()) = 0;
// returns true if the track is allowed to be added to the thread.
virtual bool isTrackAllowed_l(
audio_channel_mask_t channelMask, audio_format_t format, audio_session_t sessionId,
- uid_t uid) const = 0;
+ uid_t uid) const REQUIRES(mutex()) = 0;
virtual bool supportsHapticPlayback() const = 0;
- virtual void setDownStreamPatch(const struct audio_patch* patch) = 0;
+ virtual void setDownStreamPatch(const struct audio_patch* patch)
+ EXCLUDES_ThreadBase_Mutex = 0;
- virtual IAfTrack* getTrackById_l(audio_port_handle_t trackId) = 0;
+ virtual IAfTrack* getTrackById_l(audio_port_handle_t trackId) REQUIRES(mutex()) = 0;
virtual bool hasMixer() const = 0;
virtual status_t setRequestedLatencyMode(audio_latency_mode_t mode) = 0;
- virtual status_t getSupportedLatencyModes(std::vector<audio_latency_mode_t>* modes) = 0;
+ virtual status_t getSupportedLatencyModes(std::vector<audio_latency_mode_t>* modes)
+ EXCLUDES_ThreadBase_Mutex = 0;
virtual status_t setBluetoothVariableLatencyEnabled(bool enabled) = 0;
- virtual void setStandby() = 0;
- virtual void setStandby_l() = 0;
- virtual bool waitForHalStart() = 0;
+ virtual void setStandby() EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void setStandby_l() REQUIRES(mutex()) = 0;
+ virtual bool waitForHalStart() EXCLUDES_ThreadBase_Mutex = 0;
virtual bool hasFastMixer() const = 0;
virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const = 0;
@@ -494,9 +541,9 @@
const sp<IAfThreadCallback>& afThreadCallback, IAfPlaybackThread* mainThread,
audio_io_handle_t id, bool systemReady);
- virtual void addOutputTrack(IAfPlaybackThread* thread) = 0;
+ virtual void addOutputTrack(IAfPlaybackThread* thread) EXCLUDES_ThreadBase_Mutex = 0;
virtual uint32_t waitTimeMs() const = 0;
- virtual void removeOutputTrack(IAfPlaybackThread* thread) = 0;
+ virtual void removeOutputTrack(IAfPlaybackThread* thread) EXCLUDES_ThreadBase_Mutex = 0;
};
class IAfRecordThread : public virtual IAfThreadBase {
@@ -521,32 +568,39 @@
status_t* status /*non-NULL*/,
audio_port_handle_t portId,
int32_t maxSharedAudioHistoryMs)
- REQUIRES(audio_utils::AudioFlinger_Mutex) = 0;
- virtual void destroyTrack_l(const sp<IAfRecordTrack>& track) = 0;
- virtual void removeTrack_l(const sp<IAfRecordTrack>& track) = 0;
+ REQUIRES(audio_utils::AudioFlinger_Mutex) EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void destroyTrack_l(const sp<IAfRecordTrack>& track) REQUIRES(mutex()) = 0;
+ virtual void removeTrack_l(const sp<IAfRecordTrack>& track) REQUIRES(mutex()) = 0;
virtual status_t start(
IAfRecordTrack* recordTrack, AudioSystem::sync_event_t event,
- audio_session_t triggerSession) = 0;
+ audio_session_t triggerSession) EXCLUDES_ThreadBase_Mutex = 0;
// ask the thread to stop the specified track, and
// return true if the caller should then do it's part of the stopping process
- virtual bool stop(IAfRecordTrack* recordTrack) = 0;
+ virtual bool stop(IAfRecordTrack* recordTrack) EXCLUDES_ThreadBase_Mutex = 0;
+ // NO_THREAD_SAFETY_ANALYSIS: consider atomics
virtual AudioStreamIn* getInput() const = 0;
virtual AudioStreamIn* clearInput() = 0;
virtual status_t getActiveMicrophones(
- std::vector<media::MicrophoneInfoFw>* activeMicrophones) const = 0;
- virtual status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) = 0;
- virtual status_t setPreferredMicrophoneFieldDimension(float zoom) = 0;
+ std::vector<media::MicrophoneInfoFw>* activeMicrophones)
+ const EXCLUDES_ThreadBase_Mutex = 0;
+ virtual status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction)
+ EXCLUDES_ThreadBase_Mutex = 0;
+ virtual status_t setPreferredMicrophoneFieldDimension(float zoom)
+ EXCLUDES_ThreadBase_Mutex = 0;
- virtual void addPatchTrack(const sp<IAfPatchRecord>& record) = 0;
- virtual void deletePatchTrack(const sp<IAfPatchRecord>& record) = 0;
+ virtual void addPatchTrack(const sp<IAfPatchRecord>& record)
+ EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void deletePatchTrack(const sp<IAfPatchRecord>& record)
+ EXCLUDES_ThreadBase_Mutex = 0;
virtual bool fastTrackAvailable() const = 0;
virtual void setFastTrackAvailable(bool available) = 0;
- virtual void setRecordSilenced(audio_port_handle_t portId, bool silenced) = 0;
+ virtual void setRecordSilenced(audio_port_handle_t portId, bool silenced)
+ EXCLUDES_ThreadBase_Mutex = 0;
virtual bool hasFastCapture() const = 0;
virtual void checkBtNrec() = 0;
@@ -555,8 +609,8 @@
virtual status_t shareAudioHistory(
const std::string& sharedAudioPackageName,
audio_session_t sharedSessionId = AUDIO_SESSION_NONE,
- int64_t sharedAudioStartMs = -1) = 0;
- virtual void resetAudioHistory_l() = 0;
+ int64_t sharedAudioStartMs = -1) EXCLUDES_ThreadBase_Mutex = 0;
+ virtual void resetAudioHistory_l() REQUIRES(mutex()) = 0;
};
class IAfMmapThread : public virtual IAfThreadBase {
@@ -576,7 +630,7 @@
const sp<MmapStreamCallback>& callback,
audio_port_handle_t deviceId,
audio_port_handle_t portId) = 0;
- virtual void disconnect() = 0;
+ virtual void disconnect() EXCLUDES_ThreadBase_Mutex = 0;
// MmapStreamInterface handling (see adapter)
virtual status_t createMmapBuffer(
@@ -591,10 +645,12 @@
virtual status_t reportData(const void* buffer, size_t frameCount) = 0;
// TODO(b/291317898) move to IAfThreadBase?
- virtual void invalidateTracks(std::set<audio_port_handle_t>& portIds) = 0;
+ virtual void invalidateTracks(std::set<audio_port_handle_t>& portIds)
+ EXCLUDES_ThreadBase_Mutex = 0;
// Sets the UID records silence - TODO(b/291317898) move to IAfMmapCaptureThread
- virtual void setRecordSilenced(audio_port_handle_t portId, bool silenced) = 0;
+ virtual void setRecordSilenced(audio_port_handle_t portId, bool silenced)
+ EXCLUDES_ThreadBase_Mutex = 0;
virtual sp<IAfMmapPlaybackThread> asIAfMmapPlaybackThread() { return nullptr; }
virtual sp<IAfMmapCaptureThread> asIAfMmapCaptureThread() { return nullptr; }
@@ -606,7 +662,7 @@
const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
AudioHwDevice* hwDev, AudioStreamOut* output, bool systemReady);
- virtual AudioStreamOut* clearOutput() = 0;
+ virtual AudioStreamOut* clearOutput() EXCLUDES_ThreadBase_Mutex = 0;
};
class IAfMmapCaptureThread : public virtual IAfMmapThread {
@@ -615,7 +671,7 @@
const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
AudioHwDevice* hwDev, AudioStreamIn* input, bool systemReady);
- virtual AudioStreamIn* clearInput() = 0;
+ virtual AudioStreamIn* clearInput() EXCLUDES_ThreadBase_Mutex = 0;
};
} // namespace android
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 0b73fba..1fa5262 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -875,7 +875,7 @@
} break;
case CFG_EVENT_IO: {
IoConfigEventData *data = (IoConfigEventData *)event->mData.get();
- ioConfigChanged(data->mEvent, data->mPid, data->mPortId);
+ ioConfigChanged_l(data->mEvent, data->mPid, data->mPortId);
} break;
case CFG_EVENT_SET_PARAMETER: {
SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get();
@@ -886,22 +886,22 @@
}
} break;
case CFG_EVENT_CREATE_AUDIO_PATCH: {
- const DeviceTypeSet oldDevices = getDeviceTypes();
+ const DeviceTypeSet oldDevices = getDeviceTypes_l();
CreateAudioPatchConfigEventData *data =
(CreateAudioPatchConfigEventData *)event->mData.get();
event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle);
- const DeviceTypeSet newDevices = getDeviceTypes();
+ const DeviceTypeSet newDevices = getDeviceTypes_l();
configChanged = oldDevices != newDevices;
mLocalLog.log("CFG_EVENT_CREATE_AUDIO_PATCH: old device %s (%s) new device %s (%s)",
dumpDeviceTypes(oldDevices).c_str(), toString(oldDevices).c_str(),
dumpDeviceTypes(newDevices).c_str(), toString(newDevices).c_str());
} break;
case CFG_EVENT_RELEASE_AUDIO_PATCH: {
- const DeviceTypeSet oldDevices = getDeviceTypes();
+ const DeviceTypeSet oldDevices = getDeviceTypes_l();
ReleaseAudioPatchConfigEventData *data =
(ReleaseAudioPatchConfigEventData *)event->mData.get();
event->mStatus = releaseAudioPatch_l(data->mHandle);
- const DeviceTypeSet newDevices = getDeviceTypes();
+ const DeviceTypeSet newDevices = getDeviceTypes_l();
configChanged = oldDevices != newDevices;
mLocalLog.log("CFG_EVENT_RELEASE_AUDIO_PATCH: old device %s (%s) new device %s (%s)",
dumpDeviceTypes(oldDevices).c_str(), toString(oldDevices).c_str(),
@@ -1089,9 +1089,9 @@
}
// Note: output device may be used by capture threads for effects such as AEC.
dprintf(fd, " Output devices: %s (%s)\n",
- dumpDeviceTypes(outDeviceTypes()).c_str(), toString(outDeviceTypes()).c_str());
+ dumpDeviceTypes(outDeviceTypes_l()).c_str(), toString(outDeviceTypes_l()).c_str());
dprintf(fd, " Input device: %#x (%s)\n",
- inDeviceType(), toString(inDeviceType()).c_str());
+ inDeviceType_l(), toString(inDeviceType_l()).c_str());
dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, toString(mAudioSource).c_str());
// Dump timestamp statistics for the Thread types that support it.
@@ -1102,7 +1102,8 @@
|| mType == OFFLOAD
|| mType == SPATIALIZER) {
dprintf(fd, " Timestamp stats: %s\n", mTimestampVerifier.toString().c_str());
- dprintf(fd, " Timestamp corrected: %s\n", isTimestampCorrectionEnabled() ? "yes" : "no");
+ dprintf(fd, " Timestamp corrected: %s\n",
+ isTimestampCorrectionEnabled_l() ? "yes" : "no");
}
if (mLastIoBeginNs > 0) { // MMAP may not set this
@@ -1979,7 +1980,7 @@
}
template <typename T>
-void ThreadBase::ActiveTracks<T>::updatePowerState(
+void ThreadBase::ActiveTracks<T>::updatePowerState_l(
const sp<ThreadBase>& thread, bool force) {
// Updates ActiveTracks client uids to the thread wakelock.
if (mActiveTracksGeneration != mLastActiveTracksGeneration || force) {
@@ -2045,6 +2046,7 @@
// Call only from threadLoop() or when it is idle.
// Do not call from high performance code as this may do binder rpc to the MediaMetrics service.
void ThreadBase::sendStatistics(bool force)
+NO_THREAD_SAFETY_ANALYSIS
{
// Do not log if we have no stats.
// We choose the timestamp verifier because it is the most likely item to be present.
@@ -2076,8 +2078,8 @@
item->setInt64(MM_PREFIX "channelMask", (int64_t)mChannelMask);
item->setCString(MM_PREFIX "encoding", toString(mFormat).c_str());
item->setInt32(MM_PREFIX "frameCount", (int32_t)mFrameCount);
- item->setCString(MM_PREFIX "outDevice", toString(outDeviceTypes()).c_str());
- item->setCString(MM_PREFIX "inDevice", toString(inDeviceType()).c_str());
+ item->setCString(MM_PREFIX "outDevice", toString(outDeviceTypes_l()).c_str());
+ item->setCString(MM_PREFIX "inDevice", toString(inDeviceType_l()).c_str());
// thread statistics
if (mIoJitterMs.getN() > 0) {
@@ -2791,6 +2793,8 @@
return latency_l();
}
uint32_t PlaybackThread::latency_l() const
+NO_THREAD_SAFETY_ANALYSIS
+// Fix later.
{
uint32_t latency;
if (initCheck() == NO_ERROR && mOutput->stream->getLatency(&latency) == OK) {
@@ -3026,7 +3030,7 @@
return mOutput->stream->selectPresentation(presentationId, programId);
}
-void PlaybackThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
+void PlaybackThread::ioConfigChanged_l(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId) {
ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event);
sp<AudioIoDescriptor> desc;
@@ -3048,7 +3052,7 @@
desc = sp<AudioIoDescriptor>::make(mId);
break;
}
- mAfThreadCallback->ioConfigChanged(event, desc, pid);
+ mAfThreadCallback->ioConfigChanged_l(event, desc, pid);
}
void PlaybackThread::onWriteReady()
@@ -3234,8 +3238,8 @@
if (hasMixer()) {
mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
}
- ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount,
- mNormalFrameCount);
+ ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames",
+ (size_t)mFrameCount, mNormalFrameCount);
// Check if we want to throttle the processing to no more than 2x normal rate
mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */);
@@ -3472,7 +3476,7 @@
ALOGD("ro.audio.silent is ignored since no output device is set");
return;
}
- if (isSingleDeviceType(outDeviceTypes(), AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) {
+ if (isSingleDeviceType(outDeviceTypes_l(), AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) {
ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX");
return;
}
@@ -3642,7 +3646,7 @@
// make sure standby delay is not too short when connected to an A2DP sink to avoid
// truncating audio when going to standby.
- if (!Intersection(outDeviceTypes(), getAudioDeviceOutAllA2dpSet()).empty()) {
+ if (!Intersection(outDeviceTypes_l(), getAudioDeviceOutAllA2dpSet()).empty()) {
if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) {
mStandbyDelayNs = kDefaultStandbyTimeInNsecs;
}
@@ -3990,7 +3994,7 @@
// If the device is AUDIO_DEVICE_OUT_BUS, check for downstream latency.
//
// Note: we access outDeviceTypes() outside of mutex().
- if (isMsdDevice() && outDeviceTypes().count(AUDIO_DEVICE_OUT_BUS) != 0) {
+ if (isMsdDevice() && outDeviceTypes_l().count(AUDIO_DEVICE_OUT_BUS) != 0) {
// Here, we try for the AF lock, but do not block on it as the latency
// is more informational.
if (mAfThreadCallback->mutex().try_lock()) {
@@ -4135,7 +4139,7 @@
// mMixerStatusIgnoringFastTracks is also updated internally
mMixerStatus = prepareTracks_l(&tracksToRemove);
- mActiveTracks.updatePowerState(this);
+ mActiveTracks.updatePowerState_l(this);
metadataUpdate = updateMetadata_l();
@@ -4508,9 +4512,10 @@
// notify of throttle end on debug log
// but prevent spamming for bluetooth
ALOGD_IF(!isSingleDeviceType(
- outDeviceTypes(), audio_is_a2dp_out_device) &&
+ outDeviceTypes_l(), audio_is_a2dp_out_device) &&
!isSingleDeviceType(
- outDeviceTypes(), audio_is_hearing_aid_out_device),
+ outDeviceTypes_l(),
+ audio_is_hearing_aid_out_device),
"mixer(%p) throttle end: throttle time(%u)", this, diff);
mThreadThrottleEndMs = mThreadThrottleTimeMs;
}
@@ -4606,7 +4611,7 @@
timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
mSampleRate);
- if (isTimestampCorrectionEnabled()) {
+ if (isTimestampCorrectionEnabled_l()) {
ALOGVV("TS_BEFORE: %d %lld %lld", id(),
(long long)timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
(long long)timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]);
@@ -7218,6 +7223,7 @@
{
if (mFlushPending || mHwPaused) {
// If a flush is pending or track was paused, just discard buffered data
+ audio_utils::lock_guard l(mutex());
flushHw_l();
} else {
mMixerStatus = MIXER_DRAIN_ALL;
@@ -8261,7 +8267,7 @@
}
- mActiveTracks.updatePowerState(this);
+ mActiveTracks.updatePowerState_l(this);
updateMetadata_l();
@@ -8439,7 +8445,12 @@
mTimestampVerifier.add(position, time, mSampleRate);
// Correct timestamps
- if (isTimestampCorrectionEnabled()) {
+ bool timestampCorrectionEnabled = false;
+ {
+ audio_utils::lock_guard l(mutex());
+ timestampCorrectionEnabled = isTimestampCorrectionEnabled_l();
+ }
+ if (timestampCorrectionEnabled) {
ALOGVV("TS_BEFORE: %d %lld %lld",
id(), (long long)time, (long long)position);
auto correctedTimestamp = mTimestampVerifier.getLastCorrectedTimestamp();
@@ -9418,7 +9429,7 @@
{
// disable AEC and NS if the device is a BT SCO headset supporting those
// pre processings
- bool suspend = audio_is_bluetooth_sco_device(inDeviceType()) &&
+ bool suspend = audio_is_bluetooth_sco_device(inDeviceType_l()) &&
mAfThreadCallback->btNrecIsOff();
if (mBtNrecSuspended.exchange(suspend) != suspend) {
for (size_t i = 0; i < mEffectChains.size(); i++) {
@@ -9529,7 +9540,7 @@
return {};
}
-void RecordThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
+void RecordThread::ioConfigChanged_l(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId) {
sp<AudioIoDescriptor> desc;
switch (event) {
@@ -9547,7 +9558,7 @@
desc = sp<AudioIoDescriptor>::make(mId);
break;
}
- mAfThreadCallback->ioConfigChanged(event, desc, pid);
+ mAfThreadCallback->ioConfigChanged_l(event, desc, pid);
}
void RecordThread::readInputParameters_l()
@@ -10345,7 +10356,10 @@
bool MmapThread::threadLoop()
{
- checkSilentMode_l();
+ {
+ audio_utils::unique_lock _l(mutex());
+ checkSilentMode_l();
+ }
const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
@@ -10385,7 +10399,7 @@
checkInvalidTracks_l();
- mActiveTracks.updatePowerState(this);
+ mActiveTracks.updatePowerState_l(this);
updateMetadata_l();
@@ -10442,7 +10456,7 @@
return {};
}
-void MmapThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
+void MmapThread::ioConfigChanged_l(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId __unused) {
sp<AudioIoDescriptor> desc;
bool isInput = false;
@@ -10464,7 +10478,7 @@
desc = sp<AudioIoDescriptor>::make(mId);
break;
}
- mAfThreadCallback->ioConfigChanged(event, desc, pid);
+ mAfThreadCallback->ioConfigChanged_l(event, desc, pid);
}
status_t MmapThread::createAudioPatch_l(const struct audio_patch* patch,
@@ -10760,14 +10774,24 @@
AudioHwDevice *hwDev, AudioStreamOut *output, bool systemReady)
: MmapThread(afThreadCallback, id, hwDev, output->stream, systemReady, true /* isOut */),
mStreamType(AUDIO_STREAM_MUSIC),
- mStreamVolume(1.0),
- mStreamMute(false),
mOutput(output)
{
snprintf(mThreadName, kThreadNameLength, "AudioMmapOut_%X", id);
mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
mMasterVolume = afThreadCallback->masterVolume_l();
mMasterMute = afThreadCallback->masterMute_l();
+
+ for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
+ const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
+ mStreamTypes[stream].volume = 0.0f;
+ mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
+ }
+ // Audio patch and call assistant volume are always max
+ mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
+ mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
+ mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].volume = 1.0f;
+ mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
+
if (mAudioHwDev) {
if (mAudioHwDev->canSetMasterVolume()) {
mMasterVolume = 1.0;
@@ -10824,8 +10848,8 @@
void MmapPlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
{
audio_utils::lock_guard _l(mutex());
+ mStreamTypes[stream].volume = value;
if (stream == mStreamType) {
- mStreamVolume = value;
broadcast_l();
}
}
@@ -10833,17 +10857,14 @@
float MmapPlaybackThread::streamVolume(audio_stream_type_t stream) const
{
audio_utils::lock_guard _l(mutex());
- if (stream == mStreamType) {
- return mStreamVolume;
- }
- return 0.0f;
+ return mStreamTypes[stream].volume;
}
void MmapPlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
{
audio_utils::lock_guard _l(mutex());
+ mStreamTypes[stream].mute = muted;
if (stream == mStreamType) {
- mStreamMute= muted;
broadcast_l();
}
}
@@ -10883,14 +10904,13 @@
{
float volume;
- if (mMasterMute || mStreamMute) {
+ if (mMasterMute || streamMuted_l()) {
volume = 0;
} else {
- volume = mMasterVolume * mStreamVolume;
+ volume = mMasterVolume * streamVolume_l();
}
if (volume != mHalVolFloat) {
-
// Convert volumes from float to 8.24
uint32_t vol = (uint32_t)(volume * (1 << 24));
@@ -10924,8 +10944,8 @@
track->setMetadataHasChanged();
track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
/*muteState=*/{mMasterMute,
- mStreamVolume == 0.f,
- mStreamMute,
+ streamVolume_l() == 0.f,
+ streamMuted_l(),
// TODO(b/241533526): adjust logic to include mute from AppOps
false /*muteFromPlaybackRestricted*/,
false /*muteFromClientVolume*/,
@@ -11038,7 +11058,7 @@
MmapThread::dumpInternals_l(fd, args);
dprintf(fd, " Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d\n",
- mStreamType, mStreamVolume, mHalVolFloat, mStreamMute);
+ mStreamType, streamVolume_l(), mHalVolFloat, streamMuted_l());
dprintf(fd, " Master volume: %f Master mute %d\n", mMasterVolume, mMasterMute);
}
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 3105ad7..9020342 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -51,7 +51,7 @@
~ThreadBase() override;
status_t readyToRun() final;
- void clearPowerManager() final;
+ void clearPowerManager() final EXCLUDES_ThreadBase_Mutex;
// base for record and playback
enum {
@@ -68,11 +68,9 @@
class ConfigEventData: public RefBase {
public:
- virtual ~ConfigEventData() {}
-
virtual void dump(char *buffer, size_t size) = 0;
protected:
- ConfigEventData() {}
+ ConfigEventData() = default;
};
// Config event sequence by client if status needed (e.g binder thread calling setParameters()):
@@ -103,14 +101,22 @@
}
}
- audio_utils::mutex& mutex() const { return mMutex; }
+ audio_utils::mutex& mutex() const RETURN_CAPABILITY(audio_utils::ConfigEvent_Mutex) {
+ return mMutex;
+ }
const int mType; // event type e.g. CFG_EVENT_IO
mutable audio_utils::mutex mMutex; // mutex associated with mCondition
audio_utils::condition_variable mCondition; // condition for status return
+
+ // NO_THREAD_SAFETY_ANALYSIS Can we add GUARDED_BY?
status_t mStatus; // status communicated to sender
- bool mWaitStatus; // true if sender is waiting for status
- bool mRequiresSystemReady; // true if must wait for system ready to enter event queue
- sp<ConfigEventData> mData; // event specific parameter data
+
+ bool mWaitStatus GUARDED_BY(mutex()); // true if sender is waiting for status
+ // true if must wait for system ready to enter event queue
+ bool mRequiresSystemReady GUARDED_BY(mutex());
+
+ // NO_THREAD_SAFETY_ANALYSIS Can we add GUARDED_BY?
+ sp<ConfigEventData> mData; // event specific parameter data
protected:
explicit ConfigEvent(int type, bool requiresSystemReady = false) :
@@ -197,7 +203,7 @@
}
const struct audio_patch mPatch;
- audio_patch_handle_t mHandle;
+ audio_patch_handle_t mHandle; // cannot be const
};
class CreateAudioPatchConfigEvent : public ConfigEvent {
@@ -219,7 +225,7 @@
snprintf(buffer, size, "- Patch handle: %u\n", mHandle);
}
- audio_patch_handle_t mHandle;
+ const audio_patch_handle_t mHandle;
};
class ReleaseAudioPatchConfigEvent : public ConfigEvent {
@@ -240,7 +246,7 @@
snprintf(buffer, size, "- Devices: %s", android::toString(mOutDevices).c_str());
}
- DeviceDescriptorBaseVector mOutDevices;
+ const DeviceDescriptorBaseVector mOutDevices;
};
class UpdateOutDevicesConfigEvent : public ConfigEvent {
@@ -260,7 +266,7 @@
snprintf(buffer, size, "- mMaxSharedAudioHistoryMs: %d", mMaxSharedAudioHistoryMs);
}
- int32_t mMaxSharedAudioHistoryMs;
+ const int32_t mMaxSharedAudioHistoryMs;
};
class ResizeBufferConfigEvent : public ConfigEvent {
@@ -289,15 +295,14 @@
class PMDeathRecipient : public IBinder::DeathRecipient {
public:
explicit PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
- virtual ~PMDeathRecipient() {}
// IBinder::DeathRecipient
- virtual void binderDied(const wp<IBinder>& who);
+ void binderDied(const wp<IBinder>& who) final;
private:
DISALLOW_COPY_AND_ASSIGN(PMDeathRecipient);
- wp<ThreadBase> mThread;
+ const wp<ThreadBase> mThread;
};
type_t type() const final { return mType; }
@@ -311,8 +316,9 @@
uint32_t channelCount() const final { return mChannelCount; }
audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
uint32_t hapticChannelCount() const override { return 0; }
- uint32_t latency_l() const override { return 0; }
- void setVolumeForOutput_l(float /* left */, float /* right */) const override {}
+ uint32_t latency_l() const override { return 0; } // NO_THREAD_SAFETY_ANALYSIS
+ void setVolumeForOutput_l(float /* left */, float /* right */) const override
+ REQUIRES(mutex()) {}
// Return's the HAL's frame count i.e. fast mixer buffer size.
size_t frameCountHAL() const final { return mFrameCount; }
@@ -320,44 +326,49 @@
// Should be "virtual status_t requestExitAndWait()" and override same
// method in Thread, but Thread::requestExitAndWait() is not yet virtual.
- void exit() final;
- status_t setParameters(const String8& keyValuePairs) final;
+ void exit() final EXCLUDES_ThreadBase_Mutex;
+ status_t setParameters(const String8& keyValuePairs) final EXCLUDES_ThreadBase_Mutex;
// sendConfigEvent_l() must be called with ThreadBase::mutex() held
// Can temporarily release the lock if waiting for a reply from
// processConfigEvents_l().
- status_t sendConfigEvent_l(sp<ConfigEvent>& event);
+ status_t sendConfigEvent_l(sp<ConfigEvent>& event) REQUIRES(mutex());
void sendIoConfigEvent(audio_io_config_event_t event, pid_t pid = 0,
- audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final EXCLUDES_ThreadBase_Mutex;
void sendIoConfigEvent_l(audio_io_config_event_t event, pid_t pid = 0,
- audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
- void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) final;
- void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp) final;
- status_t sendSetParameterConfigEvent_l(const String8& keyValuePair) final;
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final REQUIRES(mutex());
+ void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) final
+ EXCLUDES_ThreadBase_Mutex;
+ void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp) final
+ REQUIRES(mutex());
+ status_t sendSetParameterConfigEvent_l(const String8& keyValuePair) final REQUIRES(mutex());
status_t sendCreateAudioPatchConfigEvent(const struct audio_patch* patch,
- audio_patch_handle_t* handle) final;
- status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle) final;
+ audio_patch_handle_t* handle) final EXCLUDES_ThreadBase_Mutex;
+ status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle) final
+ EXCLUDES_ThreadBase_Mutex;
status_t sendUpdateOutDeviceConfigEvent(
- const DeviceDescriptorBaseVector& outDevices) final;
- void sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs) final;
- void sendCheckOutputStageEffectsEvent() final;
- void sendCheckOutputStageEffectsEvent_l() final;
- void sendHalLatencyModesChangedEvent_l() final;
+ const DeviceDescriptorBaseVector& outDevices) final EXCLUDES_ThreadBase_Mutex;
+ void sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs) final REQUIRES(mutex());
+ void sendCheckOutputStageEffectsEvent() final EXCLUDES_ThreadBase_Mutex;
+ void sendCheckOutputStageEffectsEvent_l() final REQUIRES(mutex());
+ void sendHalLatencyModesChangedEvent_l() final REQUIRES(mutex());
- void processConfigEvents_l() final;
+ void processConfigEvents_l() final REQUIRES(mutex());
void setCheckOutputStageEffects() override {}
void updateOutDevices(const DeviceDescriptorBaseVector& outDevices) override;
void toAudioPortConfig(struct audio_port_config* config) override;
- void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) override;
+ void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) override REQUIRES(mutex());
// see note at declaration of mStandby, mOutDevice and mInDevice
bool inStandby() const override { return mStandby; }
- const DeviceTypeSet outDeviceTypes() const final {
+ const DeviceTypeSet outDeviceTypes_l() const final REQUIRES(mutex()) {
return getAudioDeviceTypes(mOutDeviceTypeAddrs);
}
- audio_devices_t inDeviceType() const final { return mInDeviceTypeAddr.mType; }
- DeviceTypeSet getDeviceTypes() const final {
- return isOutput() ? outDeviceTypes() : DeviceTypeSet({inDeviceType()});
+ audio_devices_t inDeviceType_l() const final REQUIRES(mutex()) {
+ return mInDeviceTypeAddr.mType;
+ }
+ DeviceTypeSet getDeviceTypes_l() const final REQUIRES(mutex()) {
+ return isOutput() ? outDeviceTypes_l() : DeviceTypeSet({inDeviceType_l()});
}
const AudioDeviceTypeAddrVector& outDeviceTypeAddrs() const final {
@@ -417,16 +428,19 @@
// ThreadBase mutex before processing the mixer and effects. This guarantees the
// integrity of the chains during the process.
// Also sets the parameter 'effectChains' to current value of mEffectChains.
- void lockEffectChains_l(Vector<sp<IAfEffectChain>>& effectChains) final;
+ void lockEffectChains_l(Vector<sp<IAfEffectChain>>& effectChains) final REQUIRES(mutex());
// unlock effect chains after process
void unlockEffectChains(const Vector<sp<IAfEffectChain>>& effectChains) final;
// get a copy of mEffectChains vector
- Vector<sp<IAfEffectChain>> getEffectChains_l() const final { return mEffectChains; };
+ Vector<sp<IAfEffectChain>> getEffectChains_l() const final REQUIRES(mutex()) {
+ return mEffectChains;
+ }
// set audio mode to all effect chains
void setMode(audio_mode_t mode) final;
// get effect module with corresponding ID on specified audio session
sp<IAfEffectModule> getEffect(audio_session_t sessionId, int effectId) const final;
- sp<IAfEffectModule> getEffect_l(audio_session_t sessionId, int effectId) const final;
+ sp<IAfEffectModule> getEffect_l(audio_session_t sessionId, int effectId) const final
+ REQUIRES(mutex());
// add and effect module. Also creates the effect chain is none exists for
// the effects audio session. Only called in a context of moving an effect
// from one thread to another
@@ -434,17 +448,18 @@
REQUIRES(audio_utils::AudioFlinger_Mutex, mutex());
// remove and effect module. Also removes the effect chain is this was the last
// effect
- void removeEffect_l(const sp<IAfEffectModule>& effect, bool release = false) final;
+ void removeEffect_l(const sp<IAfEffectModule>& effect, bool release = false) final
+ REQUIRES(mutex());
// disconnect an effect handle from module and destroy module if last handle
void disconnectEffectHandle(IAfEffectHandle* handle, bool unpinIfLast) final;
// detach all tracks connected to an auxiliary effect
- void detachAuxEffect_l(int /* effectId */) override {}
+ void detachAuxEffect_l(int /* effectId */) override REQUIRES(mutex()) {}
// TODO(b/291317898) - remove hasAudioSession_l below.
- uint32_t hasAudioSession_l(audio_session_t sessionId) const override = 0;
- uint32_t hasAudioSession(audio_session_t sessionId) const final {
- std::lock_guard _l(mutex());
- return hasAudioSession_l(sessionId);
- }
+ uint32_t hasAudioSession_l(audio_session_t sessionId) const override REQUIRES(mutex()) = 0;
+ uint32_t hasAudioSession(audio_session_t sessionId) const final EXCLUDES_ThreadBase_Mutex {
+ std::lock_guard _l(mutex());
+ return hasAudioSession_l(sessionId);
+ }
template <typename T>
uint32_t hasAudioSession_l(audio_session_t sessionId, const T& tracks) const {
@@ -476,9 +491,9 @@
// the value returned by default implementation is not important as the
// strategy is only meaningful for PlaybackThread which implements this method
product_strategy_t getStrategyForSession_l(
- audio_session_t /* sessionId */) const override {
- return static_cast<product_strategy_t>(0);
- }
+ audio_session_t /* sessionId */) const override REQUIRES(mutex()){
+ return static_cast<product_strategy_t>(0);
+ }
// check if some effects must be suspended/restored when an effect is enabled
// or disabled
@@ -497,38 +512,40 @@
sp<IMemory> pipeMemory() const override { return nullptr; }
- void systemReady() final;
+ void systemReady() final EXCLUDES_ThreadBase_Mutex;
- void broadcast_l() final;
+ void broadcast_l() final REQUIRES(mutex());
- bool isTimestampCorrectionEnabled() const override { return false; }
+ bool isTimestampCorrectionEnabled_l() const override REQUIRES(mutex()) { return false; }
bool isMsdDevice() const final { return mIsMsdDevice; }
void dump(int fd, const Vector<String16>& args) override;
// deliver stats to mediametrics.
- void sendStatistics(bool force) final;
+ void sendStatistics(bool force) final EXCLUDES_ThreadBase_Mutex;
audio_utils::mutex& mutex() const final RETURN_CAPABILITY(audio_utils::ThreadBase_Mutex) {
return mMutex;
}
mutable audio_utils::mutex mMutex;
- void onEffectEnable(const sp<IAfEffectModule>& effect) final;
- void onEffectDisable() final;
+ void onEffectEnable(const sp<IAfEffectModule>& effect) final EXCLUDES_ThreadBase_Mutex;
+ void onEffectDisable() final EXCLUDES_ThreadBase_Mutex;
// invalidateTracksForAudioSession_l must be called with holding mutex().
- void invalidateTracksForAudioSession_l(audio_session_t /* sessionId */) const override {}
+ void invalidateTracksForAudioSession_l(audio_session_t /* sessionId */) const override
+ REQUIRES(mutex()) {}
// Invalidate all the tracks with the given audio session.
- void invalidateTracksForAudioSession(audio_session_t sessionId) const final {
+ void invalidateTracksForAudioSession(audio_session_t sessionId) const final
+ EXCLUDES_ThreadBase_Mutex {
std::lock_guard _l(mutex());
invalidateTracksForAudioSession_l(sessionId);
}
template <typename T>
- void invalidateTracksForAudioSession_l(audio_session_t sessionId,
- const T& tracks) const {
+ void invalidateTracksForAudioSession_l(audio_session_t sessionId,
+ const T& tracks) const REQUIRES(mutex()) {
for (size_t i = 0; i < tracks.size(); ++i) {
const sp<IAfTrackBase>& track = tracks[i];
if (sessionId == track->sessionId()) {
@@ -553,41 +570,41 @@
effect_uuid_t mType; // effect type UUID
};
- void acquireWakeLock();
- virtual void acquireWakeLock_l();
- void releaseWakeLock();
- void releaseWakeLock_l();
- void updateWakeLockUids_l(const SortedVector<uid_t> &uids);
- void getPowerManager_l();
+ void acquireWakeLock() EXCLUDES_ThreadBase_Mutex;
+ virtual void acquireWakeLock_l() REQUIRES(mutex());
+ void releaseWakeLock() EXCLUDES_ThreadBase_Mutex;
+ void releaseWakeLock_l() REQUIRES(mutex());
+ void updateWakeLockUids_l(const SortedVector<uid_t> &uids) REQUIRES(mutex());
+ void getPowerManager_l() REQUIRES(mutex());
// suspend or restore effects of the specified type (or all if type is NULL)
// on a given session. The number of suspend requests is counted and restore
// occurs when all suspend requests are cancelled.
- void setEffectSuspended_l(const effect_uuid_t *type,
+ void setEffectSuspended_l(const effect_uuid_t *type,
bool suspend,
- audio_session_t sessionId) final;
+ audio_session_t sessionId) final REQUIRES(mutex());
// updated mSuspendedSessions when an effect is suspended or restored
- void updateSuspendedSessions_l(const effect_uuid_t *type,
+ void updateSuspendedSessions_l(const effect_uuid_t *type,
bool suspend,
- audio_session_t sessionId);
+ audio_session_t sessionId) REQUIRES(mutex());
// check if some effects must be suspended when an effect chain is added
- void checkSuspendOnAddEffectChain_l(const sp<IAfEffectChain>& chain);
+ void checkSuspendOnAddEffectChain_l(const sp<IAfEffectChain>& chain) REQUIRES(mutex());
// sends the metadata of the active tracks to the HAL
struct MetadataUpdate {
std::vector<playback_track_metadata_v7_t> playbackMetadataUpdate;
std::vector<record_track_metadata_v7_t> recordMetadataUpdate;
};
- virtual MetadataUpdate updateMetadata_l() = 0;
+ virtual MetadataUpdate updateMetadata_l() REQUIRES(mutex()) = 0;
String16 getWakeLockTag();
- virtual void preExit() { }
- virtual void setMasterMono_l(bool mono __unused) { }
+ virtual void preExit() EXCLUDES_ThreadBase_Mutex {}
+ virtual void setMasterMono_l(bool mono __unused) REQUIRES(mutex()) {}
virtual bool requireMonoBlend() { return false; }
// called within the threadLoop to obtain timestamp from the HAL.
- virtual status_t threadloop_getHalTimestamp_l(
- ExtendedTimestamp *timestamp __unused) const {
+ virtual status_t threadloop_getHalTimestamp_l(
+ ExtendedTimestamp *timestamp __unused) const REQUIRES(mutex()) {
return INVALID_OPERATION;
}
public:
@@ -595,11 +612,12 @@
product_strategy_t getStrategyForStream(audio_stream_type_t stream) const;
protected:
- virtual void onHalLatencyModesChanged_l() {}
+ virtual void onHalLatencyModesChanged_l() REQUIRES(mutex()) {}
- virtual void dumpInternals_l(int fd __unused, const Vector<String16>& args __unused)
- { }
- virtual void dumpTracks_l(int fd __unused, const Vector<String16>& args __unused) { }
+ virtual void dumpInternals_l(int fd __unused, const Vector<String16>& args __unused)
+ REQUIRES(mutex()) {}
+ virtual void dumpTracks_l(int fd __unused, const Vector<String16>& args __unused)
+ REQUIRES(mutex()) {}
const type_t mType;
@@ -624,8 +642,10 @@
// HAL format if Fastmixer is used.
audio_format_t mHALFormat;
size_t mBufferSize; // HAL buffer size for read() or write()
- AudioDeviceTypeAddrVector mOutDeviceTypeAddrs; // output device types and addresses
- AudioDeviceTypeAddr mInDeviceTypeAddr; // input device type and address
+
+ // output device types and addresses
+ AudioDeviceTypeAddrVector mOutDeviceTypeAddrs GUARDED_BY(mutex());
+ AudioDeviceTypeAddr mInDeviceTypeAddr GUARDED_BY(mutex()); // input device type and address
Vector< sp<ConfigEvent> > mConfigEvents;
Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready
@@ -764,7 +784,8 @@
// ThreadBase thread.
void clear();
// periodically called in the threadLoop() to update power state uids.
- void updatePowerState(const sp<ThreadBase>& thread, bool force = false);
+ void updatePowerState_l(const sp<ThreadBase>& thread, bool force = false)
+ REQUIRES(audio_utils::ThreadBase_Mutex);
/** @return true if one or move active tracks was added or removed since the
* last time this function was called or the vector was created.
@@ -799,11 +820,11 @@
bool mHasChanged = false;
};
- SimpleLog mLocalLog;
+ SimpleLog mLocalLog; // locked internally
private:
- void dumpBase_l(int fd, const Vector<String16>& args);
- void dumpEffectChains_l(int fd, const Vector<String16>& args);
+ void dumpBase_l(int fd, const Vector<String16>& args) REQUIRES(mutex());
+ void dumpEffectChains_l(int fd, const Vector<String16>& args) REQUIRES(mutex());
};
// --- PlaybackThread ---
@@ -836,15 +857,15 @@
~PlaybackThread() override;
// Thread virtuals
- bool threadLoop() final;
+ bool threadLoop() final EXCLUDES_ThreadBase_Mutex;
// RefBase
void onFirstRef() override;
status_t checkEffectCompatibility_l(
- const effect_descriptor_t* desc, audio_session_t sessionId) final;
+ const effect_descriptor_t* desc, audio_session_t sessionId) final REQUIRES(mutex());
- void addOutputTrack_l(const sp<IAfTrack>& track) final {
+ void addOutputTrack_l(const sp<IAfTrack>& track) final REQUIRES(mutex()) {
mTracks.add(track);
}
@@ -877,9 +898,9 @@
protected:
virtual bool waitingAsyncCallback();
- virtual bool waitingAsyncCallback_l();
- virtual bool shouldStandby_l();
- virtual void onAddNewTrack_l();
+ virtual bool waitingAsyncCallback_l() REQUIRES(mutex());
+ virtual bool shouldStandby_l() REQUIRES(mutex());
+ virtual void onAddNewTrack_l() REQUIRES(mutex());
public: // AsyncCallbackThread
void onAsyncError(); // error reported by AsyncCallbackThread
protected:
@@ -888,20 +909,20 @@
const std::basic_string<uint8_t>& metadataBs) final;
// ThreadBase virtuals
- virtual void preExit();
+ void preExit() final EXCLUDES_ThreadBase_Mutex;
virtual bool keepWakeLock() const { return true; }
- virtual void acquireWakeLock_l() {
+ virtual void acquireWakeLock_l() REQUIRES(mutex()) {
ThreadBase::acquireWakeLock_l();
- mActiveTracks.updatePowerState(this, true /* force */);
+ mActiveTracks.updatePowerState_l(this, true /* force */);
}
virtual void checkOutputStageEffects() {}
virtual void setHalLatencyMode_l() {}
- void dumpInternals_l(int fd, const Vector<String16>& args) override;
- void dumpTracks_l(int fd, const Vector<String16>& args) final;
+ void dumpInternals_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
+ void dumpTracks_l(int fd, const Vector<String16>& args) final REQUIRES(mutex());
public:
@@ -910,15 +931,15 @@
// return estimated latency in milliseconds, as reported by HAL
uint32_t latency() const final;
// same, but lock must already be held
- uint32_t latency_l() const final;
+ uint32_t latency_l() const final /* REQUIRES(mutex()) */; // NO_THREAD_SAFETY_ANALYSIS
// VolumeInterface
void setMasterVolume(float value) final;
- void setMasterBalance(float balance) override;
+ void setMasterBalance(float balance) override EXCLUDES_ThreadBase_Mutex;
void setMasterMute(bool muted) final;
- void setStreamVolume(audio_stream_type_t stream, float value) final;
- void setStreamMute(audio_stream_type_t stream, bool muted) final;
- float streamVolume(audio_stream_type_t stream) const final;
+ void setStreamVolume(audio_stream_type_t stream, float value) final EXCLUDES_ThreadBase_Mutex;
+ void setStreamMute(audio_stream_type_t stream, bool muted) final EXCLUDES_ThreadBase_Mutex;
+ float streamVolume(audio_stream_type_t stream) const final EXCLUDES_ThreadBase_Mutex;
void setVolumeForOutput_l(float left, float right) const final;
sp<IAfTrack> createTrack_l(
@@ -949,7 +970,7 @@
return mActiveTracks.indexOf(track) >= 0;
}
- AudioStreamOut* getOutput_l() const final { return mOutput; }
+ AudioStreamOut* getOutput_l() const final REQUIRES(mutex()) { return mOutput; }
AudioStreamOut* getOutput() const final;
AudioStreamOut* clearOutput() final;
sp<StreamHalInterface> stream() const final;
@@ -967,38 +988,48 @@
bool isSuspended() const final
{ return android_atomic_acquire_load(&mSuspended) > 0; }
- String8 getParameters(const String8& keys);
- void ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
+ String8 getParameters(const String8& keys) EXCLUDES_ThreadBase_Mutex;
+
+ // Hold either the AudioFlinger::mutex or the ThreadBase::mutex
+ void ioConfigChanged_l(audio_io_config_event_t event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
- status_t getRenderPosition(uint32_t* halFrames, uint32_t* dspFrames) const final;
+ status_t getRenderPosition(uint32_t* halFrames, uint32_t* dspFrames) const final
+ EXCLUDES_ThreadBase_Mutex;
// Consider also removing and passing an explicit mMainBuffer initialization
// parameter to AF::IAfTrack::Track().
float* sinkBuffer() const final {
return reinterpret_cast<float *>(mSinkBuffer); };
- void detachAuxEffect_l(int effectId) final;
+ void detachAuxEffect_l(int effectId) final REQUIRES(mutex());
- status_t attachAuxEffect(const sp<IAfTrack>& track, int EffectId) final;
- status_t attachAuxEffect_l(const sp<IAfTrack>& track, int EffectId) final;
+ status_t attachAuxEffect(const sp<IAfTrack>& track, int EffectId) final
+ EXCLUDES_ThreadBase_Mutex;
+ status_t attachAuxEffect_l(const sp<IAfTrack>& track, int EffectId) final REQUIRES(mutex());
- status_t addEffectChain_l(const sp<IAfEffectChain>& chain) final;
- size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) final;
- uint32_t hasAudioSession_l(audio_session_t sessionId) const final {
+ status_t addEffectChain_l(const sp<IAfEffectChain>& chain) final REQUIRES(mutex());
+ size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) final REQUIRES(mutex());
+ uint32_t hasAudioSession_l(audio_session_t sessionId) const final REQUIRES(mutex()) {
return ThreadBase::hasAudioSession_l(sessionId, mTracks);
}
- product_strategy_t getStrategyForSession_l(audio_session_t sessionId) const final;
+ product_strategy_t getStrategyForSession_l(audio_session_t sessionId) const final
+ REQUIRES(mutex());
- status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) final;
+ status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) final
+ EXCLUDES_ThreadBase_Mutex;
+ // could be static.
bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const final;
- // called with AudioFlinger lock held
- bool invalidateTracks_l(audio_stream_type_t streamType) final;
- bool invalidateTracks_l(std::set<audio_port_handle_t>& portIds) final;
+ // Does this require the AudioFlinger mutex as well?
+ bool invalidateTracks_l(audio_stream_type_t streamType) final
+ REQUIRES(mutex());
+ bool invalidateTracks_l(std::set<audio_port_handle_t>& portIds) final
+ REQUIRES(mutex());
void invalidateTracks(audio_stream_type_t streamType) override;
// Invalidate tracks by a set of port ids. The port id will be removed from
// the given set if the corresponding track is found and invalidated.
- void invalidateTracks(std::set<audio_port_handle_t>& portIds) override;
+ void invalidateTracks(std::set<audio_port_handle_t>& portIds) override
+ EXCLUDES_ThreadBase_Mutex;
size_t frameCount() const final { return mNormalFrameCount; }
@@ -1006,30 +1037,32 @@
return mMixerChannelMask;
}
- status_t getTimestamp_l(AudioTimestamp& timestamp) final;
+ status_t getTimestamp_l(AudioTimestamp& timestamp) final REQUIRES(mutex());
- void addPatchTrack(const sp<IAfPatchTrack>& track) final;
- void deletePatchTrack(const sp<IAfPatchTrack>& track) final;
+ void addPatchTrack(const sp<IAfPatchTrack>& track) final EXCLUDES_ThreadBase_Mutex;
+ void deletePatchTrack(const sp<IAfPatchTrack>& track) final EXCLUDES_ThreadBase_Mutex;
+ // NO_THREAD_SAFETY_ANALYSIS - fix this to use atomics.
void toAudioPortConfig(struct audio_port_config* config) final;
// Return the asynchronous signal wait time.
- int64_t computeWaitTimeNs_l() const override { return INT64_MAX; }
+ int64_t computeWaitTimeNs_l() const override REQUIRES(mutex()) { return INT64_MAX; }
// returns true if the track is allowed to be added to the thread.
bool isTrackAllowed_l(
audio_channel_mask_t channelMask __unused,
audio_format_t format __unused,
audio_session_t sessionId __unused,
- uid_t uid) const override {
+ uid_t uid) const override REQUIRES(mutex()) {
return trackCountForUid_l(uid) < PlaybackThread::kMaxTracksPerUid
&& mTracks.size() < PlaybackThread::kMaxTracks;
}
- bool isTimestampCorrectionEnabled() const final {
- return audio_is_output_devices(mTimestampCorrectedDevice)
- && outDeviceTypes().count(mTimestampCorrectedDevice) != 0;
+ bool isTimestampCorrectionEnabled_l() const final REQUIRES(mutex()) {
+ return audio_is_output_devices(mTimestampCorrectedDevice)
+ && outDeviceTypes_l().count(mTimestampCorrectedDevice) != 0;
}
+ // NO_THREAD_SAFETY_ANALYSIS - fix this to be atomic.
bool isStreamInitialized() const final {
return !(mOutput == nullptr || mOutput->stream == nullptr);
}
@@ -1046,12 +1079,12 @@
return (mHapticChannelMask & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE;
}
- void setDownStreamPatch(const struct audio_patch* patch) final {
+ void setDownStreamPatch(const struct audio_patch* patch) final EXCLUDES_ThreadBase_Mutex {
std::lock_guard _l(mutex());
mDownStreamPatch = *patch;
}
- IAfTrack* getTrackById_l(audio_port_handle_t trackId) final;
+ IAfTrack* getTrackById_l(audio_port_handle_t trackId) final REQUIRES(mutex());
bool hasMixer() const final {
return mType == MIXER || mType == DUPLICATING || mType == SPATIALIZER;
@@ -1079,14 +1112,14 @@
setStandby_l();
}
- void setStandby_l() final {
+ void setStandby_l() final REQUIRES(mutex()) {
mStandby = true;
mHalStarted = false;
mKernelPositionOnStandby =
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
}
- bool waitForHalStart() final {
+ bool waitForHalStart() final EXCLUDES_ThreadBase_Mutex {
audio_utils::unique_lock _l(mutex());
static const nsecs_t kWaitHalTimeoutNs = seconds(2);
nsecs_t endWaitTimetNs = systemTime() + kWaitHalTimeoutNs;
@@ -1224,24 +1257,28 @@
// No sleep in standby mode; waits on a condition
// Code snippets that are temporarily lifted up out of threadLoop() until the merge
- virtual void checkSilentMode_l() final; // consider unification with MMapThread
+
+ // consider unification with MMapThread
+ virtual void checkSilentMode_l() final REQUIRES(mutex());
// Non-trivial for DUPLICATING only
virtual void saveOutputTracks() { }
virtual void clearOutputTracks() { }
// Cache various calculated values, at threadLoop() entry and after a parameter change
- virtual void cacheParameters_l();
+ virtual void cacheParameters_l() REQUIRES(mutex());
void setCheckOutputStageEffects() override {
mCheckOutputStageEffects.store(true);
}
- virtual uint32_t correctLatency_l(uint32_t latency) const;
+ virtual uint32_t correctLatency_l(uint32_t latency) const REQUIRES(mutex());
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
- audio_patch_handle_t *handle);
- virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle);
+ audio_patch_handle_t *handle) REQUIRES(mutex());
+ virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle)
+ REQUIRES(mutex());
+ // NO_THREAD_SAFETY_ANALYSIS - fix this to use atomics
bool usesHwAvSync() const final { return mType == DIRECT && mOutput != nullptr
&& mHwSupportsPause
&& (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); }
@@ -1249,22 +1286,23 @@
uint32_t trackCountForUid_l(uid_t uid) const;
void invalidateTracksForAudioSession_l(
- audio_session_t sessionId) const override {
+ audio_session_t sessionId) const override REQUIRES(mutex()) {
ThreadBase::invalidateTracksForAudioSession_l(sessionId, mTracks);
}
DISALLOW_COPY_AND_ASSIGN(PlaybackThread);
- status_t addTrack_l(const sp<IAfTrack>& track) final;
- bool destroyTrack_l(const sp<IAfTrack>& track) final;
+ status_t addTrack_l(const sp<IAfTrack>& track) final REQUIRES(mutex());
+ bool destroyTrack_l(const sp<IAfTrack>& track) final REQUIRES(mutex());
- void removeTrack_l(const sp<IAfTrack>& track);
+ void removeTrack_l(const sp<IAfTrack>& track) REQUIRES(mutex());
- void readOutputParameters_l();
- MetadataUpdate updateMetadata_l() final;
- virtual void sendMetadataToBackend_l(const StreamOutHalInterface::SourceMetadata& metadata);
+ void readOutputParameters_l() REQUIRES(mutex());
+ MetadataUpdate updateMetadata_l() final REQUIRES(mutex());
+ virtual void sendMetadataToBackend_l(const StreamOutHalInterface::SourceMetadata& metadata)
+ REQUIRES(mutex()) ;
- void collectTimestamps_l();
+ void collectTimestamps_l() REQUIRES(mutex());
// The Tracks class manages tracks added and removed from the Thread.
template <typename T>
@@ -1412,7 +1450,7 @@
protected:
// accessed by both binder threads and within threadLoop(), lock on mutex needed
- uint32_t& fastTrackAvailMask_l() final { return mFastTrackAvailMask; }
+ uint32_t& fastTrackAvailMask_l() final REQUIRES(mutex()) { return mFastTrackAvailMask; }
uint32_t mFastTrackAvailMask; // bit i set if fast track [i] is available
bool mHwSupportsPause;
bool mHwPaused;
@@ -1481,18 +1519,19 @@
// Thread virtuals
- bool checkForNewParameter_l(const String8& keyValuePair, status_t& status) final;
+ bool checkForNewParameter_l(const String8& keyValuePair, status_t& status) final
+ REQUIRES(mutex());
bool isTrackAllowed_l(
audio_channel_mask_t channelMask, audio_format_t format,
- audio_session_t sessionId, uid_t uid) const final;
+ audio_session_t sessionId, uid_t uid) const final REQUIRES(mutex());
protected:
mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) override;
uint32_t idleSleepTimeUs() const final;
uint32_t suspendSleepTimeUs() const final;
- void cacheParameters_l() override;
+ void cacheParameters_l() override REQUIRES(mutex());
- void acquireWakeLock_l() final {
+ void acquireWakeLock_l() final REQUIRES(mutex()) {
PlaybackThread::acquireWakeLock_l();
if (hasFastMixer()) {
mFastMixer->setBoottimeOffset(
@@ -1500,18 +1539,19 @@
}
}
- void dumpInternals_l(int fd, const Vector<String16>& args) override;
+ void dumpInternals_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
// threadLoop snippets
ssize_t threadLoop_write() override;
void threadLoop_standby() override;
void threadLoop_mix() override;
void threadLoop_sleepTime() override;
- uint32_t correctLatency_l(uint32_t latency) const final;
+ uint32_t correctLatency_l(uint32_t latency) const final REQUIRES(mutex());
status_t createAudioPatch_l(
- const struct audio_patch* patch, audio_patch_handle_t* handle) final;
- status_t releaseAudioPatch_l(const audio_patch_handle_t handle) final;
+ const struct audio_patch* patch, audio_patch_handle_t* handle)
+ final REQUIRES(mutex());
+ status_t releaseAudioPatch_l(const audio_patch_handle_t handle) final REQUIRES(mutex());
AudioMixer* mAudioMixer; // normal mixer
@@ -1550,8 +1590,8 @@
return mFastMixerDumpState.mTracks[fastIndex].mUnderruns;
}
- status_t threadloop_getHalTimestamp_l(
- ExtendedTimestamp *timestamp) const override {
+ status_t threadloop_getHalTimestamp_l(
+ ExtendedTimestamp *timestamp) const override REQUIRES(mutex()) {
if (mNormalSink.get() != nullptr) {
return mNormalSink->getTimestamp(*timestamp);
}
@@ -1575,16 +1615,16 @@
// and blending without limiter is idempotent but inefficient to do twice.
virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); }
- void setMasterBalance(float balance) override {
+ void setMasterBalance(float balance) override EXCLUDES_ThreadBase_Mutex {
mMasterBalance.store(balance);
if (hasFastMixer()) {
mFastMixer->setMasterBalance(balance);
}
}
- void updateHalSupportedLatencyModes_l();
- void onHalLatencyModesChanged_l() override;
- void setHalLatencyMode_l() override;
+ void updateHalSupportedLatencyModes_l() REQUIRES(mutex());
+ void onHalLatencyModesChanged_l() override REQUIRES(mutex());
+ void setHalLatencyMode_l() override REQUIRES(mutex());
};
class DirectOutputThread : public PlaybackThread, public virtual IAfDirectOutputThread {
@@ -1599,35 +1639,35 @@
const audio_offload_info_t& offloadInfo)
: DirectOutputThread(afThreadCallback, output, id, DIRECT, systemReady, offloadInfo) { }
- virtual ~DirectOutputThread();
+ ~DirectOutputThread() override;
status_t selectPresentation(int presentationId, int programId) final;
// Thread virtuals
virtual bool checkForNewParameter_l(const String8& keyValuePair,
- status_t& status);
+ status_t& status) REQUIRES(mutex());
- void flushHw_l() override;
+ void flushHw_l() override REQUIRES(mutex());
- void setMasterBalance(float balance) override;
+ void setMasterBalance(float balance) override EXCLUDES_ThreadBase_Mutex;
protected:
virtual uint32_t activeSleepTimeUs() const;
virtual uint32_t idleSleepTimeUs() const;
virtual uint32_t suspendSleepTimeUs() const;
- virtual void cacheParameters_l();
+ virtual void cacheParameters_l() REQUIRES(mutex());
- void dumpInternals_l(int fd, const Vector<String16>& args) override;
+ void dumpInternals_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
// threadLoop snippets
- virtual mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove);
+ virtual mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) REQUIRES(mutex());
virtual void threadLoop_mix();
virtual void threadLoop_sleepTime();
virtual void threadLoop_exit();
- virtual bool shouldStandby_l();
+ virtual bool shouldStandby_l() REQUIRES(mutex());
- virtual void onAddNewTrack_l();
+ virtual void onAddNewTrack_l() REQUIRES(mutex());
const audio_offload_info_t mOffloadInfo;
@@ -1637,7 +1677,7 @@
DirectOutputThread(const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
audio_io_handle_t id, ThreadBase::type_t type, bool systemReady,
const audio_offload_info_t& offloadInfo);
- void processVolume_l(IAfTrack *track, bool lastTrack);
+ void processVolume_l(IAfTrack *track, bool lastTrack) REQUIRES(mutex());
bool isTunerStream() const { return (mOffloadInfo.content_id > 0); }
// prepareTracks_l() tells threadLoop_mix() the name of the single active track
@@ -1652,7 +1692,7 @@
public:
virtual bool hasFastMixer() const { return false; }
- virtual int64_t computeWaitTimeNs_l() const override;
+ virtual int64_t computeWaitTimeNs_l() const override REQUIRES(mutex());
status_t threadloop_getHalTimestamp_l(ExtendedTimestamp *timestamp) const override {
// For DIRECT and OFFLOAD threads, query the output sink directly.
@@ -1679,19 +1719,20 @@
audio_io_handle_t id, bool systemReady,
const audio_offload_info_t& offloadInfo);
virtual ~OffloadThread() {};
- void flushHw_l() override;
+ void flushHw_l() final REQUIRES(mutex());
protected:
// threadLoop snippets
- virtual mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove);
- virtual void threadLoop_exit();
+ mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) final
+ REQUIRES(mutex());
+ void threadLoop_exit() final;
- virtual bool waitingAsyncCallback();
- virtual bool waitingAsyncCallback_l();
- virtual void invalidateTracks(audio_stream_type_t streamType);
- void invalidateTracks(std::set<audio_port_handle_t>& portIds) override;
+ bool waitingAsyncCallback() final;
+ bool waitingAsyncCallback_l() final REQUIRES(mutex());
+ void invalidateTracks(audio_stream_type_t streamType) final EXCLUDES_ThreadBase_Mutex;
+ void invalidateTracks(std::set<audio_port_handle_t>& portIds) final EXCLUDES_ThreadBase_Mutex;
- virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); }
+ bool keepWakeLock() const final { return (mKeepWakeLock || (mDrainSequence & 1)); }
private:
size_t mPausedWriteLength; // length in bytes of write interrupted by pause
@@ -1730,7 +1771,9 @@
mutable audio_utils::mutex mMutex;
bool mAsyncError;
- audio_utils::mutex& mutex() const { return mMutex; }
+ audio_utils::mutex& mutex() const RETURN_CAPABILITY(audio_utils::AsyncCallbackThread_Mutex) {
+ return mMutex;
+ }
};
class DuplicatingThread : public MixerThread, public IAfDuplicatingThread {
@@ -1745,32 +1788,32 @@
}
// Thread virtuals
- void addOutputTrack(IAfPlaybackThread* thread) final;
- void removeOutputTrack(IAfPlaybackThread* thread) final;
+ void addOutputTrack(IAfPlaybackThread* thread) final EXCLUDES_ThreadBase_Mutex;
+ void removeOutputTrack(IAfPlaybackThread* thread) final EXCLUDES_ThreadBase_Mutex;
uint32_t waitTimeMs() const final { return mWaitTimeMs; }
void sendMetadataToBackend_l(
- const StreamOutHalInterface::SourceMetadata& metadata) override;
+ const StreamOutHalInterface::SourceMetadata& metadata) final REQUIRES(mutex());
protected:
virtual uint32_t activeSleepTimeUs() const;
- void dumpInternals_l(int fd, const Vector<String16>& args) override;
+ void dumpInternals_l(int fd, const Vector<String16>& args) final REQUIRES(mutex());
private:
bool outputsReady();
protected:
// threadLoop snippets
- virtual void threadLoop_mix();
- virtual void threadLoop_sleepTime();
- virtual ssize_t threadLoop_write();
- virtual void threadLoop_standby();
- virtual void cacheParameters_l();
+ void threadLoop_mix() final;
+ void threadLoop_sleepTime() final;
+ ssize_t threadLoop_write() final;
+ void threadLoop_standby() final;
+ void cacheParameters_l() final REQUIRES(mutex());
private:
// called from threadLoop, addOutputTrack, removeOutputTrack
- virtual void updateWaitTime_l();
+ void updateWaitTime_l() REQUIRES(mutex());
protected:
- virtual void saveOutputTracks();
- virtual void clearOutputTracks();
+ void saveOutputTracks() final;
+ void clearOutputTracks() final;
private:
uint32_t mWaitTimeMs;
@@ -1779,7 +1822,7 @@
public:
virtual bool hasFastMixer() const { return false; }
status_t threadloop_getHalTimestamp_l(
- ExtendedTimestamp *timestamp) const override {
+ ExtendedTimestamp *timestamp) const override REQUIRES(mutex()) {
if (mOutputTracks.size() > 0) {
// forward the first OutputTrack's kernel information for timestamp.
const ExtendedTimestamp trackTimestamp =
@@ -1806,11 +1849,11 @@
bool hasFastMixer() const final { return false; }
- status_t setRequestedLatencyMode(audio_latency_mode_t mode) final;
+ status_t setRequestedLatencyMode(audio_latency_mode_t mode) final EXCLUDES_ThreadBase_Mutex;
protected:
- void checkOutputStageEffects() final;
- void setHalLatencyMode_l() final;
+ void checkOutputStageEffects() final EXCLUDES_ThreadBase_Mutex;
+ void setHalLatencyMode_l() final REQUIRES(mutex());
private:
// Do not request a specific mode by default
@@ -1836,15 +1879,15 @@
~RecordThread() override;
// no addTrack_l ?
- void destroyTrack_l(const sp<IAfRecordTrack>& track) final;
- void removeTrack_l(const sp<IAfRecordTrack>& track) final;
+ void destroyTrack_l(const sp<IAfRecordTrack>& track) final REQUIRES(mutex());
+ void removeTrack_l(const sp<IAfRecordTrack>& track) final REQUIRES(mutex());
// Thread virtuals
- bool threadLoop() final;
- void preExit() final;
+ bool threadLoop() final EXCLUDES_ThreadBase_Mutex;
+ void preExit() final EXCLUDES_ThreadBase_Mutex;
// RefBase
- void onFirstRef() final;
+ void onFirstRef() final EXCLUDES_ThreadBase_Mutex;
status_t initCheck() const final { return mInput == nullptr ? NO_INIT : NO_ERROR; }
@@ -1868,15 +1911,15 @@
status_t *status /*non-NULL*/,
audio_port_handle_t portId,
int32_t maxSharedAudioHistoryMs) final
- REQUIRES(audio_utils::AudioFlinger_Mutex);
+ REQUIRES(audio_utils::AudioFlinger_Mutex) EXCLUDES_ThreadBase_Mutex;
status_t start(IAfRecordTrack* recordTrack,
AudioSystem::sync_event_t event,
- audio_session_t triggerSession) final;
+ audio_session_t triggerSession) final EXCLUDES_ThreadBase_Mutex;
// ask the thread to stop the specified track, and
// return true if the caller should then do it's part of the stopping process
- bool stop(IAfRecordTrack* recordTrack) final;
+ bool stop(IAfRecordTrack* recordTrack) final EXCLUDES_ThreadBase_Mutex;
AudioStreamIn* getInput() const final { return mInput; }
AudioStreamIn* clearInput() final;
@@ -1884,27 +1927,30 @@
virtual sp<StreamHalInterface> stream() const;
- virtual bool checkForNewParameter_l(const String8& keyValuePair,
- status_t& status);
- virtual void cacheParameters_l() {}
- virtual String8 getParameters(const String8& keys);
- void ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
+ virtual bool checkForNewParameter_l(const String8& keyValuePair,
+ status_t& status) REQUIRES(mutex());
+ virtual void cacheParameters_l() REQUIRES(mutex()) {}
+ virtual String8 getParameters(const String8& keys) EXCLUDES_ThreadBase_Mutex;
+
+ // Hold either the AudioFlinger::mutex or the ThreadBase::mutex
+ void ioConfigChanged_l(audio_io_config_event_t event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
- audio_patch_handle_t *handle);
- virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle);
- void updateOutDevices(const DeviceDescriptorBaseVector& outDevices) override;
- void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) override;
+ audio_patch_handle_t *handle) REQUIRES(mutex());
+ virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) REQUIRES(mutex());
+ void updateOutDevices(const DeviceDescriptorBaseVector& outDevices) override
+ EXCLUDES_ThreadBase_Mutex;
+ void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) override REQUIRES(mutex());
- void addPatchTrack(const sp<IAfPatchRecord>& record) final;
- void deletePatchTrack(const sp<IAfPatchRecord>& record) final;
+ void addPatchTrack(const sp<IAfPatchRecord>& record) final EXCLUDES_ThreadBase_Mutex;
+ void deletePatchTrack(const sp<IAfPatchRecord>& record) final EXCLUDES_ThreadBase_Mutex;
- void readInputParameters_l();
- uint32_t getInputFramesLost() const final;
+ void readInputParameters_l() REQUIRES(mutex());
+ uint32_t getInputFramesLost() const final EXCLUDES_ThreadBase_Mutex;
- virtual status_t addEffectChain_l(const sp<IAfEffectChain>& chain);
- virtual size_t removeEffectChain_l(const sp<IAfEffectChain>& chain);
- uint32_t hasAudioSession_l(audio_session_t sessionId) const override {
+ virtual status_t addEffectChain_l(const sp<IAfEffectChain>& chain) REQUIRES(mutex());
+ virtual size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) REQUIRES(mutex());
+ uint32_t hasAudioSession_l(audio_session_t sessionId) const override REQUIRES(mutex()) {
return ThreadBase::hasAudioSession_l(sessionId, mTracks);
}
@@ -1913,7 +1959,8 @@
// FIXME replace by Set [and implement Bag/Multiset for other uses].
KeyedVector<audio_session_t, bool> sessionIds() const;
- status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) override;
+ status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) override
+ EXCLUDES_ThreadBase_Mutex;
bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const override;
static void syncStartEventCallback(const wp<audioflinger::SyncEvent>& event);
@@ -1922,52 +1969,55 @@
bool hasFastCapture() const final { return mFastCapture != 0; }
virtual void toAudioPortConfig(struct audio_port_config *config);
- virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc,
- audio_session_t sessionId);
+ virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc,
+ audio_session_t sessionId) REQUIRES(mutex());
- virtual void acquireWakeLock_l() {
+ virtual void acquireWakeLock_l() REQUIRES(mutex()) {
ThreadBase::acquireWakeLock_l();
- mActiveTracks.updatePowerState(this, true /* force */);
+ mActiveTracks.updatePowerState_l(this, true /* force */);
}
- void checkBtNrec() final;
+ void checkBtNrec() final EXCLUDES_ThreadBase_Mutex;
// Sets the UID records silence
- void setRecordSilenced(audio_port_handle_t portId, bool silenced) final;
+ void setRecordSilenced(audio_port_handle_t portId, bool silenced) final
+ EXCLUDES_ThreadBase_Mutex;
status_t getActiveMicrophones(
- std::vector<media::MicrophoneInfoFw>* activeMicrophones) const final;
- status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) final;
- status_t setPreferredMicrophoneFieldDimension(float zoom) final;
+ std::vector<media::MicrophoneInfoFw>* activeMicrophones) const final
+ EXCLUDES_ThreadBase_Mutex;
+ status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) final
+ EXCLUDES_ThreadBase_Mutex;
+ status_t setPreferredMicrophoneFieldDimension(float zoom) final EXCLUDES_ThreadBase_Mutex;
- MetadataUpdate updateMetadata_l() override;
+ MetadataUpdate updateMetadata_l() override REQUIRES(mutex());
bool fastTrackAvailable() const final { return mFastTrackAvail; }
void setFastTrackAvailable(bool available) final { mFastTrackAvail = available; }
- bool isTimestampCorrectionEnabled() const override {
+ bool isTimestampCorrectionEnabled_l() const override REQUIRES(mutex()) {
// checks popcount for exactly one device.
// Is currently disabled. Before enabling,
// verify compressed record timestamps.
return audio_is_input_device(mTimestampCorrectedDevice)
- && inDeviceType() == mTimestampCorrectedDevice;
+ && inDeviceType_l() == mTimestampCorrectedDevice;
}
status_t shareAudioHistory(const std::string& sharedAudioPackageName,
audio_session_t sharedSessionId = AUDIO_SESSION_NONE,
- int64_t sharedAudioStartMs = -1) final;
+ int64_t sharedAudioStartMs = -1) final EXCLUDES_ThreadBase_Mutex;
status_t shareAudioHistory_l(const std::string& sharedAudioPackageName,
audio_session_t sharedSessionId = AUDIO_SESSION_NONE,
- int64_t sharedAudioStartMs = -1);
- void resetAudioHistory_l() final;
+ int64_t sharedAudioStartMs = -1) REQUIRES(mutex());
+ void resetAudioHistory_l() final REQUIRES(mutex());
bool isStreamInitialized() const final {
return !(mInput == nullptr || mInput->stream == nullptr);
}
protected:
- void dumpInternals_l(int fd, const Vector<String16>& args) override;
- void dumpTracks_l(int fd, const Vector<String16>& args) override;
+ void dumpInternals_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
+ void dumpTracks_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
private:
// Enter standby if not already in standby, and set mStandby flag
@@ -1976,10 +2026,10 @@
// Call the HAL standby method unconditionally, and don't change mStandby flag
void inputStandBy();
- void checkBtNrec_l();
+ void checkBtNrec_l() REQUIRES(mutex());
- int32_t getOldestFront_l();
- void updateFronts_l(int32_t offset);
+ int32_t getOldestFront_l() REQUIRES(mutex());
+ void updateFronts_l(int32_t offset) REQUIRES(mutex());
AudioStreamIn *mInput;
Source *mSource;
@@ -2067,16 +2117,16 @@
audio_port_handle_t deviceId,
audio_port_handle_t portId) override;
- void disconnect() final;
+ void disconnect() final EXCLUDES_ThreadBase_Mutex;
// MmapStreamInterface for adapter.
status_t createMmapBuffer(int32_t minSizeFrames, struct audio_mmap_buffer_info* info) final;
status_t getMmapPosition(struct audio_mmap_position* position) const override;
status_t start(const AudioClient& client,
const audio_attributes_t *attr,
- audio_port_handle_t* handle) final;
- status_t stop(audio_port_handle_t handle) final;
- status_t standby() final;
+ audio_port_handle_t* handle) final EXCLUDES_ThreadBase_Mutex;
+ status_t stop(audio_port_handle_t handle) final EXCLUDES_ThreadBase_Mutex;
+ status_t standby() final EXCLUDES_ThreadBase_Mutex;
status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) const = 0;
status_t reportData(const void* buffer, size_t frameCount) override;
@@ -2084,69 +2134,77 @@
void onFirstRef() final;
// Thread virtuals
- bool threadLoop() final;
+ bool threadLoop() final EXCLUDES_ThreadBase_Mutex;
// Not in ThreadBase
virtual void threadLoop_exit() final;
virtual void threadLoop_standby() final;
- virtual bool shouldStandby_l() final { return false; }
+ virtual bool shouldStandby_l() final REQUIRES(mutex()){ return false; }
virtual status_t exitStandby_l() REQUIRES(mutex());
status_t initCheck() const final { return mHalStream == nullptr ? NO_INIT : NO_ERROR; }
size_t frameCount() const final { return mFrameCount; }
- bool checkForNewParameter_l(const String8& keyValuePair, status_t& status) final;
- String8 getParameters(const String8& keys) final;
- void ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
- audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
- void readHalParameters_l();
- void cacheParameters_l() final {}
+ bool checkForNewParameter_l(const String8& keyValuePair, status_t& status)
+ final REQUIRES(mutex());
+ String8 getParameters(const String8& keys) final EXCLUDES_ThreadBase_Mutex;
+ void ioConfigChanged_l(audio_io_config_event_t event, pid_t pid = 0,
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final
+ /* holds either AF::mutex or TB::mutex */;
+ void readHalParameters_l() REQUIRES(mutex());
+ void cacheParameters_l() final REQUIRES(mutex()) {}
status_t createAudioPatch_l(
- const struct audio_patch* patch, audio_patch_handle_t* handle) final;
- status_t releaseAudioPatch_l(const audio_patch_handle_t handle) final;
+ const struct audio_patch* patch, audio_patch_handle_t* handle) final
+ REQUIRES(mutex());
+ status_t releaseAudioPatch_l(const audio_patch_handle_t handle) final
+ REQUIRES(mutex());
void toAudioPortConfig(struct audio_port_config* config) override;
sp<StreamHalInterface> stream() const final { return mHalStream; }
- status_t addEffectChain_l(const sp<IAfEffectChain>& chain) final;
- size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) final;
+ status_t addEffectChain_l(const sp<IAfEffectChain>& chain) final REQUIRES(mutex());
+ size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) final REQUIRES(mutex());
status_t checkEffectCompatibility_l(
- const effect_descriptor_t *desc, audio_session_t sessionId) final;
+ const effect_descriptor_t *desc, audio_session_t sessionId) final REQUIRES(mutex());
- uint32_t hasAudioSession_l(audio_session_t sessionId) const override {
+ uint32_t hasAudioSession_l(audio_session_t sessionId) const override REQUIRES(mutex()) {
// Note: using mActiveTracks as no mTracks here.
return ThreadBase::hasAudioSession_l(sessionId, mActiveTracks);
}
status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) final;
bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const final;
- virtual void checkSilentMode_l() {} // cannot be const (RecordThread)
- virtual void processVolume_l() {}
+ virtual void checkSilentMode_l() REQUIRES(mutex()) {} // cannot be const (RecordThread)
+ virtual void processVolume_l() REQUIRES(mutex()) {}
void checkInvalidTracks_l();
// Not in ThreadBase
virtual audio_stream_type_t streamType() const { return AUDIO_STREAM_DEFAULT; }
- virtual void invalidateTracks(audio_stream_type_t /* streamType */) {}
- void invalidateTracks(std::set<audio_port_handle_t>& /* portIds */) override {}
+ virtual void invalidateTracks(audio_stream_type_t /* streamType */)
+ EXCLUDES_ThreadBase_Mutex {}
+ void invalidateTracks(std::set<audio_port_handle_t>& /* portIds */) override
+ EXCLUDES_ThreadBase_Mutex {}
// Sets the UID records silence
void setRecordSilenced(
- audio_port_handle_t /* portId */, bool /* silenced */) override {}
+ audio_port_handle_t /* portId */, bool /* silenced */) override
+ EXCLUDES_ThreadBase_Mutex {}
bool isStreamInitialized() const override { return false; }
- void setClientSilencedState_l(audio_port_handle_t portId, bool silenced) {
+ void setClientSilencedState_l(audio_port_handle_t portId, bool silenced) REQUIRES(mutex()) {
mClientSilencedStates[portId] = silenced;
}
- size_t eraseClientSilencedState_l(audio_port_handle_t portId) {
+ size_t eraseClientSilencedState_l(audio_port_handle_t portId) REQUIRES(mutex()) {
return mClientSilencedStates.erase(portId);
}
- bool isClientSilenced_l(audio_port_handle_t portId) const {
+ bool isClientSilenced_l(audio_port_handle_t portId) const REQUIRES(mutex()) {
const auto it = mClientSilencedStates.find(portId);
return it != mClientSilencedStates.end() ? it->second : false;
}
- void setClientSilencedIfExists_l(audio_port_handle_t portId, bool silenced) {
+ void setClientSilencedIfExists_l(audio_port_handle_t portId, bool silenced)
+ REQUIRES(mutex()) {
const auto it = mClientSilencedStates.find(portId);
if (it != mClientSilencedStates.end()) {
it->second = silenced;
@@ -2154,8 +2212,8 @@
}
protected:
- void dumpInternals_l(int fd, const Vector<String16>& args) override;
- void dumpTracks_l(int fd, const Vector<String16>& args) final;
+ void dumpInternals_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
+ void dumpTracks_l(int fd, const Vector<String16>& args) final REQUIRES(mutex());
/**
* @brief mDeviceId current device port unique identifier
@@ -2195,26 +2253,27 @@
audio_port_handle_t deviceId,
audio_port_handle_t portId) final;
- AudioStreamOut* clearOutput() final;
+ AudioStreamOut* clearOutput() final EXCLUDES_ThreadBase_Mutex;
// VolumeInterface
void setMasterVolume(float value) final;
- void setMasterBalance(float /* value */) final {} // Needs implementation?
+ // Needs implementation?
+ void setMasterBalance(float /* value */) final EXCLUDES_ThreadBase_Mutex {}
void setMasterMute(bool muted) final;
- void setStreamVolume(audio_stream_type_t stream, float value) final;
- void setStreamMute(audio_stream_type_t stream, bool muted) final;
- float streamVolume(audio_stream_type_t stream) const final;
+ void setStreamVolume(audio_stream_type_t stream, float value) final EXCLUDES_ThreadBase_Mutex;
+ void setStreamMute(audio_stream_type_t stream, bool muted) final EXCLUDES_ThreadBase_Mutex;
+ float streamVolume(audio_stream_type_t stream) const final EXCLUDES_ThreadBase_Mutex;
void setMasterMute_l(bool muted) { mMasterMute = muted; }
- void invalidateTracks(audio_stream_type_t streamType) final;
- void invalidateTracks(std::set<audio_port_handle_t>& portIds) final;
+ void invalidateTracks(audio_stream_type_t streamType) final EXCLUDES_ThreadBase_Mutex;
+ void invalidateTracks(std::set<audio_port_handle_t>& portIds) final EXCLUDES_ThreadBase_Mutex;
audio_stream_type_t streamType() const final { return mStreamType; }
- void checkSilentMode_l() final;
- void processVolume_l() final;
+ void checkSilentMode_l() final REQUIRES(mutex());
+ void processVolume_l() final REQUIRES(mutex());
- MetadataUpdate updateMetadata_l() final;
+ MetadataUpdate updateMetadata_l() final REQUIRES(mutex());
void toAudioPortConfig(struct audio_port_config* config) final;
@@ -2232,13 +2291,18 @@
REQUIRES(audio_utils::AudioFlinger_Mutex);
protected:
- void dumpInternals_l(int fd, const Vector<String16>& args) final;
+ void dumpInternals_l(int fd, const Vector<String16>& args) final REQUIRES(mutex());
+ float streamVolume_l() const {
+ return mStreamTypes[mStreamType].volume;
+ }
+ bool streamMuted_l() const {
+ return mStreamTypes[mStreamType].mute;
+ }
+ stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
audio_stream_type_t mStreamType;
float mMasterVolume;
- float mStreamVolume;
bool mMasterMute;
- bool mStreamMute;
AudioStreamOut* mOutput;
mediautils::atomic_sp<audio_utils::MelProcessor> mMelProcessor;
@@ -2254,13 +2318,14 @@
return sp<IAfMmapCaptureThread>::fromExisting(this);
}
- AudioStreamIn* clearInput() final;
+ AudioStreamIn* clearInput() final EXCLUDES_ThreadBase_Mutex;
status_t exitStandby_l() REQUIRES(mutex()) final;
- MetadataUpdate updateMetadata_l() final;
- void processVolume_l() final;
- void setRecordSilenced(audio_port_handle_t portId, bool silenced) final;
+ MetadataUpdate updateMetadata_l() final REQUIRES(mutex());
+ void processVolume_l() final REQUIRES(mutex());
+ void setRecordSilenced(audio_port_handle_t portId, bool silenced) final
+ EXCLUDES_ThreadBase_Mutex;
void toAudioPortConfig(struct audio_port_config* config) final;
@@ -2281,7 +2346,7 @@
audio_io_handle_t id, bool systemReady);
protected:
- mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) final;
+ mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) final REQUIRES(mutex());
void threadLoop_mix() final;
private:
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 31246ec..4811586 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -113,8 +113,7 @@
mChannelCount(isOut ?
audio_channel_count_from_out_mask(channelMask) :
audio_channel_count_from_in_mask(channelMask)),
- mFrameSize(audio_has_proportional_frames(format) ?
- mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
+ mFrameSize(audio_bytes_per_frame(mChannelCount, format)),
mFrameCount(frameCount),
mSessionId(sessionId),
mIsOut(isOut),
@@ -451,6 +450,10 @@
if (*_aidl_return != OK) {
return Status::ok();
}
+
+ // restrict position modulo INT_MAX to avoid integer sanitization abort
+ legacy.mPosition &= INT_MAX;
+
*timestamp = legacy2aidl_AudioTimestamp_AudioTimestampInternal(legacy).value();
return Status::ok();
}
@@ -1170,9 +1173,11 @@
if (thread != 0) {
if (isOffloaded()) {
audio_utils::lock_guard _laf(thread->afThreadCallback()->mutex());
+ const bool nonOffloadableGlobalEffectEnabled =
+ thread->afThreadCallback()->isNonOffloadableGlobalEffectEnabled_l();
audio_utils::lock_guard _lth(thread->mutex());
sp<IAfEffectChain> ec = thread->getEffectChain_l(mSessionId);
- if (thread->afThreadCallback()->isNonOffloadableGlobalEffectEnabled_l() ||
+ if (nonOffloadableGlobalEffectEnabled ||
(ec != 0 && ec->isNonOffloadableEnabled())) {
invalidate();
return PERMISSION_DENIED;
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 80e098b..6c130fd 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -89,6 +89,8 @@
void importAudioPortAndPickAudioProfile(const sp<PolicyAudioPort>& policyPort,
bool force = false);
+ status_t readFromParcelable(const media::AudioPortFw& parcelable) override;
+
void setEncapsulationInfoFromHal(AudioPolicyClientInterface *clientInterface);
void dump(String8 *dst, int spaces, bool verbose = true) const;
@@ -104,7 +106,7 @@
std::string mTagName; // Unique human readable identifier for a device port found in conf file.
audio_format_t mCurrentEncodedFormat;
bool mIsDynamic = false;
- const std::string mDeclaredAddress; // Original device address
+ std::string mDeclaredAddress; // Original device address
};
class DeviceVector : public SortedVector<sp<DeviceDescriptor> >
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp
index 8c7a7de..4edd11f 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyConfig.cpp
@@ -29,6 +29,8 @@
namespace android {
+using media::audio::common::AudioDeviceAddress;
+using media::audio::common::AudioDeviceType;
using media::audio::common::AudioIoFlags;
using media::audio::common::AudioPortDeviceExt;
using media::audio::common::AudioPortExt;
@@ -113,12 +115,17 @@
ports.emplace(aidlPort.id, devicePort);
if (const auto& deviceExt = aidlPort.ext.get<AudioPortExt::device>();
- deviceExt.device.type.connection.empty()) { // Attached device
+ deviceExt.device.type.connection.empty() ||
+ // DeviceHalAidl connects remote submix input with an address.
+ (deviceExt.device.type.type == AudioDeviceType::IN_SUBMIX &&
+ deviceExt.device.address != AudioDeviceAddress())) {
+ // Attached device.
if (isInput) {
attachedInputDevices->add(devicePort);
} else {
attachedOutputDevices->add(devicePort);
- if ((deviceExt.flags & defaultDeviceFlag) != 0) {
+ if (*defaultOutputDevice == nullptr &&
+ (deviceExt.flags & defaultDeviceFlag) != 0) {
*defaultOutputDevice = devicePort;
}
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index 514601c..fe25693 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -154,6 +154,12 @@
policyPort->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
}
+status_t DeviceDescriptor::readFromParcelable(const media::AudioPortFw& parcelable) {
+ RETURN_STATUS_IF_ERROR(DeviceDescriptorBase::readFromParcelable(parcelable));
+ mDeclaredAddress = DeviceDescriptorBase::address();
+ return OK;
+}
+
void DeviceDescriptor::setEncapsulationInfoFromHal(
AudioPolicyClientInterface *clientInterface) {
AudioParameter param(String8(mDeviceTypeAddr.getAddress()));