commit 541a200502b3a06e17ddf1d04d69a70bdae1fe1a
author Eric Laurent <elaurent@google.com> Mon Jan 15 18:11:42 2024 +0100
committer Eric Laurent <elaurent@google.com> Mon Jan 22 09:39:31 2024 +0000
tree 73a2cc892db667341451beda0d8d31ceb5ab3eec
parent ebbf45e2a32afa886a9f69a6933c78b1eb131b4a

audio policy: fix call routing via audio patch

The audio source used to control the call RX path must be tagged as
internal so that the corresponding audio patch is not realized with a
software audio patch because of the lack of gain controller on sink device.

Voice call volume is controlled via setVoiceVolume() HAL API and not
track volume control or device gain control

Bug: 302519079
Test: make
Change-Id: Ie9217f1be1bf127915d6bdf42d5de0834966b9d2

services/audiopolicy/managerdefault/AudioPolicyManager.cpp

diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index a5c407a..6160f77 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -18,7 +18,7 @@
 #define LOG_TAG "APM_AudioPolicyManager"
 
 // Need to keep the log statements even in production builds
-// to enable VERBOSE logging dynamically.
+// to enable VERBOSE logging dynamicstartAudioSourceally.
 // You can enable VERBOSE logging as follows:
 // adb shell setprop log.tag.APM_AudioPolicyManager V
 #define LOG_NDEBUG 0
@@ -782,7 +782,11 @@
         .ext.device.type = AUDIO_DEVICE_IN_TELEPHONY_RX, .ext.device.address = ""
     };
     const auto aa = mEngine->getAttributesForStreamType(AUDIO_STREAM_VOICE_CALL);
-    mCallRxSourceClient = startAudioSourceInternal(&source, &aa, 0/*uid*/);
+
+    audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+    status_t status = startAudioSource(&source, &aa, &portId, 0 /*uid*/, true /*internal*/);
+    ALOGE_IF(status != OK, "%s: failed to start audio source (%d)", __func__, status);
+    mCallRxSourceClient = mAudioSources.valueFor(portId);
     ALOGE_IF(mCallRxSourceClient == nullptr,
              "%s failed to start Telephony Rx AudioSource", __func__);
 }
@@ -815,9 +819,11 @@
 
     struct audio_port_config source = {};
     srcDevice->toAudioPortConfig(&source);
-    mCallTxSourceClient = new InternalSourceClientDescriptor(
-                callTxSourceClientPortId, mUidCached, aa, source, srcDevice, sinkDevice,
-                mCommunnicationStrategy, toVolumeSource(aa));
+    mCallTxSourceClient = new SourceClientDescriptor(
+                callTxSourceClientPortId, mUidCached, aa, source, srcDevice, AUDIO_STREAM_PATCH,
+                mCommunnicationStrategy, toVolumeSource(aa), true);
+    mCallTxSourceClient->setPreferredDeviceId(sinkDevice->getId());
+
     audio_patch_handle_t patchHandle = AUDIO_PATCH_HANDLE_NONE;
     status_t status = connectAudioSourceToSink(
                 mCallTxSourceClient, sinkDevice, patchBuilder.patch(), patchHandle, mUidCached,
@@ -4874,9 +4880,11 @@
     audio_attributes_t attributes = attributes_initializer(AUDIO_USAGE_MEDIA);
     const struct audio_port_config *source = &patch->sources[0];
     sp<SourceClientDescriptor> sourceDesc =
-            new InternalSourceClientDescriptor(
-                portId, uid, attributes, *source, srcDevice, sinkDevice,
-                mEngine->getProductStrategyForAttributes(attributes), toVolumeSource(attributes));
+            new SourceClientDescriptor(
+                portId, uid, attributes, *source, srcDevice, AUDIO_STREAM_PATCH,
+                mEngine->getProductStrategyForAttributes(attributes), toVolumeSource(attributes),
+                true);
+    sourceDesc->setPreferredDeviceId(sinkDevice->getId());
 
     status_t status =
             connectAudioSourceToSink(sourceDesc, sinkDevice, patch, *handle, uid, 0 /* delayMs */);
@@ -5543,7 +5551,7 @@
 status_t AudioPolicyManager::startAudioSource(const struct audio_port_config *source,
                                               const audio_attributes_t *attributes,
                                               audio_port_handle_t *portId,
-                                              uid_t uid)
+                                              uid_t uid, bool internal)
 {
     ALOGV("%s", __FUNCTION__);
     *portId = AUDIO_PORT_HANDLE_NONE;
@@ -5576,7 +5584,7 @@
         new SourceClientDescriptor(*portId, uid, *attributes, *source, srcDevice,
                                    mEngine->getStreamTypeForAttributes(*attributes),
                                    mEngine->getProductStrategyForAttributes(*attributes),
-                                   toVolumeSource(*attributes));
+                                   toVolumeSource(*attributes), internal);
 
     status_t status = connectAudioSource(sourceDesc);
     if (status == NO_ERROR) {
@@ -5585,18 +5593,6 @@
     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());