[BUG] AudioPolicyManager: prevent unecessary force routing on SwOutput
At the begining/end of call, routing for all SwOutput is evaluated.
In case of using bridge, either the output involved in the SwBridge or
the output involved in the HwBridge for volume will be forced to be rerouted
on same device as the routing has already been evaluated for the Swoutput
hosting the call..
This CL implements this optimization.
Bug: 187173302
Test: CSV & VoIP calls
Signed-off-by: Francois Gaffie <francois.gaffie@renault.com>
Change-Id: I15102b661a4fc4afdd10888011feadf1fa7084ce
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index fb59211..2aa3be6 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -664,8 +664,8 @@
ALOGV("%s device rxDevice %s txDevice %s", __func__,
rxDevices.itemAt(0)->toString().c_str(), txSourceDevice->toString().c_str());
- disconnectTelephonyAudioSource(mCallRxSourceClientPort);
- disconnectTelephonyAudioSource(mCallTxSourceClientPort);
+ disconnectTelephonyAudioSource(mCallRxSourceClient);
+ disconnectTelephonyAudioSource(mCallTxSourceClient);
auto telephonyRxModule =
mHwModules.getModuleForDeviceType(AUDIO_DEVICE_IN_TELEPHONY_RX, AUDIO_FORMAT_DEFAULT);
@@ -745,27 +745,32 @@
void AudioPolicyManager::connectTelephonyRxAudioSource()
{
- disconnectTelephonyAudioSource(mCallRxSourceClientPort);
+ disconnectTelephonyAudioSource(mCallRxSourceClient);
const struct audio_port_config source = {
.role = AUDIO_PORT_ROLE_SOURCE, .type = AUDIO_PORT_TYPE_DEVICE,
.ext.device.type = AUDIO_DEVICE_IN_TELEPHONY_RX, .ext.device.address = ""
};
const auto aa = mEngine->getAttributesForStreamType(AUDIO_STREAM_VOICE_CALL);
- status_t status = startAudioSource(&source, &aa, &mCallRxSourceClientPort, 0/*uid*/);
- ALOGE_IF(status != NO_ERROR, "%s failed to start Telephony Rx AudioSource", __func__);
+ mCallRxSourceClient = startAudioSourceInternal(&source, &aa, 0/*uid*/);
+ ALOGE_IF(mCallRxSourceClient == nullptr,
+ "%s failed to start Telephony Rx AudioSource", __func__);
}
-void AudioPolicyManager::disconnectTelephonyAudioSource(audio_port_handle_t &portHandle)
+void AudioPolicyManager::disconnectTelephonyAudioSource(sp<SourceClientDescriptor> &clientDesc)
{
- stopAudioSource(portHandle);
- portHandle = AUDIO_PORT_HANDLE_NONE;
+ if (clientDesc == nullptr) {
+ return;
+ }
+ ALOGW_IF(stopAudioSource(clientDesc->portId()) != NO_ERROR,
+ "%s error stopping audio source", __func__);
+ clientDesc.clear();
}
void AudioPolicyManager::connectTelephonyTxAudioSource(
const sp<DeviceDescriptor> &srcDevice, const sp<DeviceDescriptor> &sinkDevice,
uint32_t delayMs)
{
- disconnectTelephonyAudioSource(mCallTxSourceClientPort);
+ disconnectTelephonyAudioSource(mCallTxSourceClient);
if (srcDevice == nullptr || sinkDevice == nullptr) {
ALOGW("%s could not create patch, invalid sink and/or source device(s)", __func__);
return;
@@ -774,19 +779,20 @@
patchBuilder.addSource(srcDevice).addSink(sinkDevice);
ALOGV("%s between source %s and sink %s", __func__,
srcDevice->toString().c_str(), sinkDevice->toString().c_str());
- mCallTxSourceClientPort = PolicyAudioPort::getNextUniqueId();
+ auto callTxSourceClientPortId = PolicyAudioPort::getNextUniqueId();
const audio_attributes_t aa = { .source = AUDIO_SOURCE_VOICE_COMMUNICATION };
struct audio_port_config source = {};
srcDevice->toAudioPortConfig(&source);
- sp<SourceClientDescriptor> sourceDesc = new InternalSourceClientDescriptor(
- mCallTxSourceClientPort, mUidCached, aa, source, srcDevice, sinkDevice,
+ mCallTxSourceClient = new InternalSourceClientDescriptor(
+ callTxSourceClientPortId, mUidCached, aa, source, srcDevice, sinkDevice,
mCommunnicationStrategy, toVolumeSource(aa));
audio_patch_handle_t patchHandle = AUDIO_PATCH_HANDLE_NONE;
status_t status = connectAudioSourceToSink(
- sourceDesc, sinkDevice, patchBuilder.patch(), patchHandle, mUidCached, delayMs);
+ mCallTxSourceClient, sinkDevice, patchBuilder.patch(), patchHandle, mUidCached,
+ delayMs);
ALOGE_IF(status != NO_ERROR, "%s() error %d creating TX audio patch", __func__, status);
if (status == NO_ERROR) {
- mAudioSources.add(mCallTxSourceClientPort, sourceDesc);
+ mAudioSources.add(callTxSourceClientPortId, mCallTxSourceClient);
}
}
@@ -855,8 +861,8 @@
rxDevices = mPrimaryOutput->devices();
}
if (oldState == AUDIO_MODE_IN_CALL) {
- disconnectTelephonyAudioSource(mCallRxSourceClientPort);
- disconnectTelephonyAudioSource(mCallTxSourceClientPort);
+ disconnectTelephonyAudioSource(mCallRxSourceClient);
+ disconnectTelephonyAudioSource(mCallTxSourceClient);
}
setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
}
@@ -866,8 +872,10 @@
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
DeviceVector newDevices = getNewOutputDevices(desc, true /*fromCache*/);
- if (state != AUDIO_MODE_IN_CALL || desc != mPrimaryOutput) {
- setOutputDevices(desc, newDevices, !newDevices.isEmpty(), 0 /*delayMs*/);
+ if (state != AUDIO_MODE_IN_CALL || (desc != mPrimaryOutput && !isTelephonyRxOrTx(desc))) {
+ bool forceRouting = !newDevices.isEmpty();
+ setOutputDevices(desc, newDevices, forceRouting, 0 /*delayMs*/, nullptr,
+ true /*requiresMuteCheck*/, !forceRouting /*requiresVolumeCheck*/);
}
}
@@ -3523,11 +3531,15 @@
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
DeviceVector newDevices = getNewOutputDevices(outputDesc, true /*fromCache*/);
- if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
+ if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) ||
+ (outputDesc != mPrimaryOutput && !isTelephonyRxOrTx(outputDesc))) {
// As done in setDeviceConnectionState, we could also fix default device issue by
// preventing the force re-routing in case of default dev that distinguishes on address.
// Let's give back to engine full device choice decision however.
- waitMs = setOutputDevices(outputDesc, newDevices, !newDevices.isEmpty(), delayMs);
+ bool forceRouting = !newDevices.isEmpty();
+ waitMs = setOutputDevices(outputDesc, newDevices, forceRouting, delayMs, nullptr,
+ true /*requiresMuteCheck*/,
+ !forceRouting /*requiresVolumeCheck*/);
// Only apply special touch sound delay once
delayMs = 0;
}
@@ -4799,6 +4811,18 @@
return status;
}
+sp<SourceClientDescriptor> AudioPolicyManager::startAudioSourceInternal(
+ const struct audio_port_config *source, const audio_attributes_t *attributes, uid_t uid)
+{
+ ALOGV("%s", __FUNCTION__);
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+
+ status_t status = startAudioSource(source, attributes, &portId, uid);
+ ALOGE_IF(status != OK, "%s: failed to start audio source (%d)", __func__, status);
+ return mAudioSources.valueFor(portId);
+}
+
+
status_t AudioPolicyManager::connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc)
{
ALOGV("%s handle %d", __FUNCTION__, sourceDesc->portId());