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frameworks/av: Include more forward-ports from TD

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Peter Cai 2023-10-08 10:42:20 -04:00
parent 4b704639a2
commit 4435ddab0a
5 changed files with 460 additions and 3 deletions
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2
frameworks/av/0001-APM-Optionally-force-load-audio-policy-for-system-si.patch
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@ -1,7 +1,7 @@
From f8b9cb1d3cdf0274414bb7d100844d2707999558 Mon Sep 17 00:00:00 2001
From: Peter Cai <peter@typeblog.net>
Date: Wed, 24 Aug 2022 15:42:39 -0400
Subject: [PATCH 1/3] APM: Optionally force-load audio policy for system-side
Subject: [PATCH 1/5] APM: Optionally force-load audio policy for system-side
bt audio HAL
Required to support our system-side bt audio implementation, i.e.

2
frameworks/av/0002-APM-Remove-A2DP-audio-ports-from-the-primary-HAL.patch
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@ -1,7 +1,7 @@
From d60a1b5fc58a7cc84b48c554cb2cffaeff14d86e Mon Sep 17 00:00:00 2001
From: Peter Cai <peter@typeblog.net>
Date: Thu, 25 Aug 2022 13:30:29 -0400
Subject: [PATCH 2/3] APM: Remove A2DP audio ports from the primary HAL
Subject: [PATCH 2/5] APM: Remove A2DP audio ports from the primary HAL
These ports defined in the primary HAL are intended for A2DP offloading,
however they do not work in general on GSIs, and will interfere with

2
frameworks/av/0003-voip-Fix-high-pitched-voice-on-Qualcomm-devices.patch
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@ -1,7 +1,7 @@
From aa1f3595b06e6d3d7027a5dd40ae183d71d032f0 Mon Sep 17 00:00:00 2001
From: ponces <ponces26@gmail.com>
Date: Mon, 24 Oct 2022 09:38:34 +0100
Subject: [PATCH 3/3] voip: Fix high pitched voice on Qualcomm devices
Subject: [PATCH 3/5] voip: Fix high pitched voice on Qualcomm devices
Change-Id: I6d314912169776b76d07d8c0301ec5249c1870a2
---

326
frameworks/av/0004-APM-Restore-S-R-and-Q-behavior-respectively-for-tele.patch Normal file
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@ -0,0 +1,326 @@
From 1a9a9faca4d5c28747a3b7d9843b67c641ed91c9 Mon Sep 17 00:00:00 2001
From: Peter Cai <peter@typeblog.net>
Date: Thu, 18 Aug 2022 15:44:46 -0400
Subject: [PATCH 4/5] APM: Restore S, R and Q behavior respectively for
telephony audio
This conditionally reverts part of b2e5cb (T), 51c9cc (S) and afd4ce (R)
when the VNDK version is equal to or before S, R and Q respectively.
On R, commit afd4ce made it so that both HW and SW bridging go through
`createAudioPatch()`, which is broken on some devices such as on MTK Q
vendor, because their HAL do not support HW patching via the newer
`createAudioPatch()` method. Instead, the patching on Q was done through
`setOutputDevices()`.
On S, commit 51c9cc refactored the related code again such that HW
bridging for the Rx direction is essentially removed, replaced with SW
bridging through `startAudioSource()`. This is, again, broken on MTK R
vendor devices.
On T, commit b2e5cb applied the same SW bridging to the Tx direction.
All of these commits rely on assumptions that are not tested through
VTS and just presumed to be true. Although we can blame MTK for not
supporting all the possible cases in their HAL, it will not fix
anything, and really frameworks code should not depend on such untested
assumptions.
To work around said issues, we restore old behavior from S, R and Q
relying on the value of `ro.vndk.version`.
Change-Id: I56d36d2aef4319935cb88a3e4771b23c6d5b2145
---
.../managerdefault/AudioPolicyManager.cpp | 206 ++++++++++++------
.../managerdefault/AudioPolicyManager.h | 3 +
2 files changed, 147 insertions(+), 62 deletions(-)
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index f093e685ba..9a90009f9e 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -689,6 +689,17 @@ status_t AudioPolicyManager::updateCallRoutingInternal(
disconnectTelephonyAudioSource(mCallRxSourceClient);
disconnectTelephonyAudioSource(mCallTxSourceClient);
+ // release existing RX patch if any
+ if (mCallRxPatch != 0) {
+ releaseAudioPatchInternal(mCallRxPatch->getHandle());
+ mCallRxPatch.clear();
+ }
+ // release TX patch if any
+ if (mCallTxPatch != 0) {
+ releaseAudioPatchInternal(mCallTxPatch->getHandle());
+ mCallTxPatch.clear();
+ }
+
auto telephonyRxModule =
mHwModules.getModuleForDeviceType(AUDIO_DEVICE_IN_TELEPHONY_RX, AUDIO_FORMAT_DEFAULT);
auto telephonyTxModule =
@@ -711,9 +722,20 @@ status_t AudioPolicyManager::updateCallRoutingInternal(
ALOGE("%s() no telephony Tx and/or RX device", __func__);
return INVALID_OPERATION;
}
- // createAudioPatchInternal now supports both HW / SW bridging
- createRxPatch = true;
- createTxPatch = true;
+ if (property_get_int32("ro.vndk.version", 31) >= 30) {
+ // createAudioPatchInternal now supports both HW / SW bridging
+ createRxPatch = true;
+ createTxPatch = true;
+ } else {
+ // pre-R behavior: some devices before VNDK 30 do not support createAudioPatch correctly
+ // for HW bridging even though they declare support for it
+ // do not create a patch (aka Sw Bridging) if Primary HW module has declared supporting a
+ // route between telephony RX to Sink device and Source device to telephony TX
+ ALOGI("%s() Using pre-R behavior for createRxPatch and createTxPatch", __func__);
+ const auto &primaryModule = telephonyRxModule;
+ createRxPatch = !primaryModule->supportsPatch(rxSourceDevice, rxDevices.itemAt(0));
+ createTxPatch = !primaryModule->supportsPatch(txSourceDevice, txSinkDevice);
+ }
} else {
// If the RX device is on the primary HW module, then use legacy routing method for
// voice calls via setOutputDevice() on primary output.
@@ -730,7 +752,14 @@ status_t AudioPolicyManager::updateCallRoutingInternal(
if (!createRxPatch) {
muteWaitMs = setOutputDevices(mPrimaryOutput, rxDevices, true, delayMs);
} else { // create RX path audio patch
- connectTelephonyRxAudioSource();
+ if (property_get_int32("ro.vndk.version", 31) >= 31) {
+ connectTelephonyRxAudioSource();
+ } else {
+ // pre-S behavior: some devices do not support SW bridging correctly when HW bridge is
+ // available through createAudioPatch(); startAudioSource() forces SW bridging.
+ ALOGI("%s() Using pre-S behavior to create HW Rx patch", __func__);
+ mCallRxPatch = createTelephonyPatch(true /*isRx*/, rxDevices.itemAt(0), delayMs);
+ }
// If the TX device is on the primary HW module but RX device is
// on other HW module, SinkMetaData of telephony input should handle it
// assuming the device uses audio HAL V5.0 and above
@@ -745,7 +774,12 @@ status_t AudioPolicyManager::updateCallRoutingInternal(
closeActiveClients(activeDesc);
}
}
- connectTelephonyTxAudioSource(txSourceDevice, txSinkDevice, delayMs);
+ if (property_get_int32("ro.vndk.version", 33) >= 33) {
+ connectTelephonyTxAudioSource(txSourceDevice, txSinkDevice, delayMs);
+ } else {
+ // pre-T behavior: hw bridging for tx too; skip the SwOutput
+ mCallTxPatch = createTelephonyPatch(false /*isRx*/, txSourceDevice, delayMs);
+ }
}
if (waitMs != nullptr) {
*waitMs = muteWaitMs;
@@ -753,6 +787,36 @@ status_t AudioPolicyManager::updateCallRoutingInternal(
return NO_ERROR;
}
+sp<AudioPatch> AudioPolicyManager::createTelephonyPatch(
+ bool isRx, const sp<DeviceDescriptor> &device, uint32_t delayMs) {
+ PatchBuilder patchBuilder;
+
+ if (device == nullptr) {
+ return nullptr;
+ }
+
+ // @TODO: still ignoring the address, or not dealing platform with multiple telephony devices
+ if (isRx) {
+ patchBuilder.addSink(device).
+ addSource(mAvailableInputDevices.getDevice(
+ AUDIO_DEVICE_IN_TELEPHONY_RX, String8(), AUDIO_FORMAT_DEFAULT));
+ } else {
+ patchBuilder.addSource(device).
+ addSink(mAvailableOutputDevices.getDevice(
+ AUDIO_DEVICE_OUT_TELEPHONY_TX, String8(), AUDIO_FORMAT_DEFAULT));
+ }
+
+ audio_patch_handle_t patchHandle = AUDIO_PATCH_HANDLE_NONE;
+ status_t status =
+ createAudioPatchInternal(patchBuilder.patch(), &patchHandle, mUidCached, delayMs, nullptr);
+ ssize_t index = mAudioPatches.indexOfKey(patchHandle);
+ if (status != NO_ERROR || index < 0) {
+ ALOGW("%s() error %d creating %s audio patch", __func__, status, isRx ? "RX" : "TX");
+ return nullptr;
+ }
+ return mAudioPatches.valueAt(index);
+}
+
bool AudioPolicyManager::isDeviceOfModule(
const sp<DeviceDescriptor>& devDesc, const char *moduleId) const {
sp<HwModule> module = mHwModules.getModuleFromName(moduleId);
@@ -4958,83 +5022,101 @@ status_t AudioPolicyManager::createAudioPatchInternal(const struct audio_patch *
// in config XML to reach the sink so that is can be declared as available.
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
sp<SwAudioOutputDescriptor> outputDesc;
- if (!sourceDesc->isInternal()) {
- // take care of dynamic routing for SwOutput selection,
- audio_attributes_t attributes = sourceDesc->attributes();
- audio_stream_type_t stream = sourceDesc->stream();
- audio_attributes_t resultAttr;
- audio_config_t config = AUDIO_CONFIG_INITIALIZER;
- config.sample_rate = sourceDesc->config().sample_rate;
- audio_channel_mask_t sourceMask = sourceDesc->config().channel_mask;
- config.channel_mask =
- (audio_channel_mask_get_representation(sourceMask)
- == AUDIO_CHANNEL_REPRESENTATION_INDEX) ? sourceMask
- : audio_channel_mask_in_to_out(sourceMask);
- config.format = sourceDesc->config().format;
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
- audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
- bool isRequestedDeviceForExclusiveUse = false;
- output_type_t outputType;
- bool isSpatialized;
- bool isBitPerfect;
- getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE, &attributes,
- &stream, sourceDesc->uid(), &config, &flags,
- &selectedDeviceId, &isRequestedDeviceForExclusiveUse,
- nullptr, &outputType, &isSpatialized, &isBitPerfect);
- if (output == AUDIO_IO_HANDLE_NONE) {
- ALOGV("%s no output for device %s",
- __FUNCTION__, sinkDevice->toString().c_str());
- return INVALID_OPERATION;
- }
- outputDesc = mOutputs.valueFor(output);
- if (outputDesc->isDuplicated()) {
- ALOGE("%s output is duplicated", __func__);
- return INVALID_OPERATION;
- }
- bool closeOutput = outputDesc->mDirectOpenCount != 0;
- sourceDesc->setSwOutput(outputDesc, closeOutput);
- } else {
- // Same for "raw patches" aka created from createAudioPatch API
- SortedVector<audio_io_handle_t> outputs =
- getOutputsForDevices(DeviceVector(sinkDevice), mOutputs);
- // if the sink device is reachable via an opened output stream, request to
- // go via this output stream by adding a second source to the patch
- // description
- output = selectOutput(outputs);
- if (output == AUDIO_IO_HANDLE_NONE) {
- ALOGE("%s no output available for internal patch sink", __func__);
- return INVALID_OPERATION;
- }
- outputDesc = mOutputs.valueFor(output);
- if (outputDesc->isDuplicated()) {
- ALOGV("%s output for device %s is duplicated",
- __func__, sinkDevice->toString().c_str());
- return INVALID_OPERATION;
+ if (sourceDesc != nullptr) {
+ if (!sourceDesc->isInternal()) {
+ // take care of dynamic routing for SwOutput selection,
+ audio_attributes_t attributes = sourceDesc->attributes();
+ audio_stream_type_t stream = sourceDesc->stream();
+ audio_attributes_t resultAttr;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = sourceDesc->config().sample_rate;
+ audio_channel_mask_t sourceMask = sourceDesc->config().channel_mask;
+ config.channel_mask =
+ (audio_channel_mask_get_representation(sourceMask)
+ == AUDIO_CHANNEL_REPRESENTATION_INDEX) ? sourceMask
+ : audio_channel_mask_in_to_out(sourceMask);
+ config.format = sourceDesc->config().format;
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ bool isRequestedDeviceForExclusiveUse = false;
+ output_type_t outputType;
+ bool isSpatialized;
+ bool isBitPerfect;
+ getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE, &attributes,
+ &stream, sourceDesc->uid(), &config, &flags,
+ &selectedDeviceId, &isRequestedDeviceForExclusiveUse,
+ nullptr, &outputType, &isSpatialized, &isBitPerfect);
+ if (output == AUDIO_IO_HANDLE_NONE) {
+ ALOGV("%s no output for device %s",
+ __FUNCTION__, sinkDevice->toString().c_str());
+ return INVALID_OPERATION;
+ }
+ outputDesc = mOutputs.valueFor(output);
+ if (outputDesc->isDuplicated()) {
+ ALOGE("%s output is duplicated", __func__);
+ return INVALID_OPERATION;
+ }
+ bool closeOutput = outputDesc->mDirectOpenCount != 0;
+ sourceDesc->setSwOutput(outputDesc, closeOutput);
+ } else {
+ // Same for "raw patches" aka created from createAudioPatch API
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevices(DeviceVector(sinkDevice), mOutputs);
+ // if the sink device is reachable via an opened output stream, request to
+ // go via this output stream by adding a second source to the patch
+ // description
+ output = selectOutput(outputs);
+ if (output == AUDIO_IO_HANDLE_NONE) {
+ ALOGE("%s no output available for internal patch sink", __func__);
+ return INVALID_OPERATION;
+ }
+ outputDesc = mOutputs.valueFor(output);
+ if (outputDesc->isDuplicated()) {
+ ALOGV("%s output for device %s is duplicated",
+ __func__, sinkDevice->toString().c_str());
+ return INVALID_OPERATION;
+ }
}
- sourceDesc->setSwOutput(outputDesc, /* closeOutput= */ false);
}
// create a software bridge in PatchPanel if:
// - source and sink devices are on different HW modules OR
// - audio HAL version is < 3.0
// - audio HAL version is >= 3.0 but no route has been declared between devices
- // - called from startAudioSource (aka sourceDesc is not internal) and source device
+ // - called from startAudioSource (aka sourceDesc is neither null nor internal) and source device
// does not have a gain controller
if (!srcDevice->hasSameHwModuleAs(sinkDevice) ||
(srcDevice->getModuleVersionMajor() < 3) ||
!srcDevice->getModule()->supportsPatch(srcDevice, sinkDevice) ||
- (!sourceDesc->isInternal() &&
+ ((sourceDesc != nullptr && !sourceDesc->isInternal()) &&
srcDevice->getAudioPort()->getGains().size() == 0)) {
// support only one sink device for now to simplify output selection logic
if (patch->num_sinks > 1) {
return INVALID_OPERATION;
}
- sourceDesc->setUseSwBridge();
+ if (sourceDesc == nullptr) {
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevices(DeviceVector(sinkDevice), mOutputs);
+ // if the sink device is reachable via an opened output stream, request to
+ // go via this output stream by adding a second source to the patch
+ // description
+ output = selectOutput(outputs);
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ outputDesc = mOutputs.valueFor(output);
+ if (outputDesc->isDuplicated()) {
+ ALOGV("%s output for device %s is duplicated",
+ __FUNCTION__, sinkDevice->toString().c_str());
+ return INVALID_OPERATION;
+ }
+ }
+ } else {
+ sourceDesc->setUseSwBridge();
+ }
if (outputDesc != nullptr) {
audio_port_config srcMixPortConfig = {};
outputDesc->toAudioPortConfig(&srcMixPortConfig, nullptr);
// for volume control, we may need a valid stream
srcMixPortConfig.ext.mix.usecase.stream =
- (!sourceDesc->isInternal() || isCallTxAudioSource(sourceDesc)) ?
+ (sourceDesc != nullptr && (!sourceDesc->isInternal() || isCallTxAudioSource(sourceDesc))) ?
mEngine->getStreamTypeForAttributes(sourceDesc->attributes()) :
AUDIO_STREAM_PATCH;
patchBuilder.addSource(srcMixPortConfig);
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 88bafefdb1..188b5732b3 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -953,6 +953,9 @@ protected:
SoundTriggerSessionCollection mSoundTriggerSessions;
+ sp<AudioPatch> mCallTxPatch;
+ sp<AudioPatch> mCallRxPatch;
+
HwAudioOutputCollection mHwOutputs;
SourceClientCollection mAudioSources;
--
2.41.0

131
frameworks/av/0005-Fix-BT-in-call-on-CAF-devices.patch Normal file
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@ -0,0 +1,131 @@
From 97421e35dd716c9b07ec44ec05265944efc96294 Mon Sep 17 00:00:00 2001
From: Pierre-Hugues Husson <phh@phh.me>
Date: Mon, 5 Aug 2019 18:09:50 +0200
Subject: [PATCH 5/5] Fix BT in-call on CAF devices
See https://github.com/phhusson/treble_experimentations/issues/374
In Qualcomm's BSP audio_policy_configuration.xml, one route is missing,
from primary output and telephony to BT SCO.
Add it if we detect telephony and bt sco, but no such route.
Change-Id: Ifea0f88276ec9a0811f3cb1973c4b06f2c82077b
---
.../managerdefinitions/src/Serializer.cpp | 93 +++++++++++++++++++
1 file changed, 93 insertions(+)
diff --git a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
index d23d55677d..4f896111cb 100644
--- a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
@@ -672,6 +672,98 @@ std::variant<status_t, RouteTraits::Element> PolicySerializer::deserialize<Route
return route;
}
+static void fixupQualcommBtScoRoute(RouteTraits::Collection& routes, DevicePortTraits::Collection& devicePorts, HwModule* ctx) {
+ // On many Qualcomm devices, there is a BT SCO Headset Mic => primary input mix
+ // But Telephony Rx => BT SCO Headset route is missing
+ // When we detect such case, add the missing route
+
+ // If we have:
+ // <route type="mix" sink="Telephony Tx" sources="voice_tx"/>
+ // <route type="mix" sink="primary input" sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
+ // <devicePort tagName="BT SCO Headset" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET" role="sink" />
+ // And no <route type="mix" sink="BT SCO Headset" />
+
+ // Add:
+ // <route type="mix" sink="BT SCO Headset" sources="primary output,deep_buffer,compressed_offload,Telephony Rx"/>
+ bool foundBtScoHeadsetDevice = false;
+ for(const auto& device: devicePorts) {
+ if(device->getTagName() == "BT SCO Headset") {
+ foundBtScoHeadsetDevice = true;
+ break;
+ }
+ }
+ if(!foundBtScoHeadsetDevice) {
+ ALOGE("No BT SCO Headset device found, don't patch policy");
+ return;
+ }
+
+ bool foundTelephony = false;
+ bool foundBtScoInput = false;
+ bool foundScoHeadsetRoute = false;
+ for(const auto& route: routes) {
+ ALOGE("Looking at route %d\n", route->getType());
+ if(route->getType() != AUDIO_ROUTE_MIX)
+ continue;
+ auto sink = route->getSink();
+ ALOGE("... With sink %s\n", sink->getTagName().c_str());
+ if(sink->getTagName() == "Telephony Tx") {
+ foundTelephony = true;
+ continue;
+ }
+ if(sink->getTagName() == "BT SCO Headset") {
+ foundScoHeadsetRoute = true;
+ break;
+ }
+ for(const auto& source: route->getSources()) {
+ ALOGE("... With source %s\n", source->getTagName().c_str());
+ if(source->getTagName() == "BT SCO Headset Mic") {
+ foundBtScoInput = true;
+ break;
+ }
+ }
+ }
+ //The route we want to add is already there
+ ALOGE("Done looking for existing routes");
+ if(foundScoHeadsetRoute)
+ return;
+
+ ALOGE("No existing route found... %d %d", foundTelephony ? 1 : 0, foundBtScoInput ? 1 : 0);
+ //We couldn't find the routes we assume are required for the function we want to add
+ if(!foundTelephony || !foundBtScoInput)
+ return;
+ ALOGE("Adding our own.");
+
+ // Add:
+ // <route type="mix" sink="BT SCO Headset" sources="primary output,deep_buffer,compressed_offload,Telephony Rx"/>
+ AudioRoute *newRoute = new AudioRoute(AUDIO_ROUTE_MIX);
+
+ auto sink = ctx->findPortByTagName("BT SCO Headset");
+ ALOGE("Got sink %p\n", sink.get());
+ newRoute->setSink(sink);
+
+ Vector<sp<PolicyAudioPort>> sources;
+ for(const auto& sourceName: {
+ "primary output",
+ "deep_buffer",
+ "compressed_offload",
+ "Telephony Rx"
+ }) {
+ auto source = ctx->findPortByTagName(sourceName);
+ ALOGE("Got source %p\n", source.get());
+ if (source.get() != nullptr) {
+ sources.add(source);
+ source->addRoute(newRoute);
+ }
+ }
+
+ newRoute->setSources(sources);
+
+ sink->addRoute(newRoute);
+
+ auto ret = routes.add(newRoute);
+ ALOGE("route add returned %zd", ret);
+}
+
template<>
std::variant<status_t, ModuleTraits::Element> PolicySerializer::deserialize<ModuleTraits>(
const xmlNode *cur, ModuleTraits::PtrSerializingCtx ctx)
@@ -739,6 +831,7 @@ std::variant<status_t, ModuleTraits::Element> PolicySerializer::deserialize<Modu
if (status != NO_ERROR) {
return status;
}
+ fixupQualcommBtScoRoute(routes, devicePorts, module.get());
module->setRoutes(routes);
for (const xmlNode *children = cur->xmlChildrenNode; children != NULL;
--
2.41.0