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-rw-r--r--include/private/media/AudioTrackShared.h1
-rw-r--r--media/codec2/sfplugin/CCodecConfig.cpp3
-rw-r--r--media/codec2/vndk/C2Store.cpp16
-rw-r--r--media/libaaudio/src/core/AAudioAudio.cpp3
-rw-r--r--media/libaaudio/src/core/AudioStream.cpp1
-rw-r--r--media/libaaudio/tests/test_various.cpp54
-rw-r--r--media/libaudioclient/AudioTrack.cpp23
-rw-r--r--media/libstagefright/ACodec.cpp5
-rw-r--r--media/libstagefright/include/media/stagefright/MediaCodecConstants.h1
-rw-r--r--media/libstagefright/rtsp/ARTPConnection.cpp1
-rw-r--r--services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp5
-rw-r--r--services/audiopolicy/managerdefault/AudioPolicyManager.cpp80
-rw-r--r--services/audiopolicy/managerdefault/AudioPolicyManager.h44
-rw-r--r--services/audiopolicy/service/AudioPolicyService.cpp3
-rw-r--r--services/audiopolicy/service/Spatializer.cpp12
15 files changed, 195 insertions, 57 deletions
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index bd6db55669..a1e1702990 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -55,6 +55,7 @@ namespace android {
// for audio_track_cblk_t::mState, to match TrackBase.h
static inline constexpr int CBLK_STATE_IDLE = 0;
+static inline constexpr int CBLK_STATE_ACTIVE = 6;
static inline constexpr int CBLK_STATE_PAUSING = 7;
/**
diff --git a/media/codec2/sfplugin/CCodecConfig.cpp b/media/codec2/sfplugin/CCodecConfig.cpp
index c2751873c2..09e24bef29 100644
--- a/media/codec2/sfplugin/CCodecConfig.cpp
+++ b/media/codec2/sfplugin/CCodecConfig.cpp
@@ -894,6 +894,9 @@ void CCodecConfig::initializeStandardParams() {
add(ConfigMapper(KEY_AAC_MAX_OUTPUT_CHANNEL_COUNT, C2_PARAMKEY_MAX_CHANNEL_COUNT, "value")
.limitTo(D::AUDIO & (D::CONFIG | D::PARAM | D::READ)));
+ add(ConfigMapper(KEY_MAX_OUTPUT_CHANNEL_COUNT, C2_PARAMKEY_MAX_CHANNEL_COUNT, "value")
+ .limitTo(D::AUDIO & (D::CONFIG | D::PARAM | D::READ)));
+
add(ConfigMapper(KEY_AAC_SBR_MODE, C2_PARAMKEY_AAC_SBR_MODE, "value")
.limitTo(D::AUDIO & D::ENCODER & (D::CONFIG | D::PARAM | D::READ))
.withMapper([](C2Value v) -> C2Value {
diff --git a/media/codec2/vndk/C2Store.cpp b/media/codec2/vndk/C2Store.cpp
index 1660c381b9..dfdd84d484 100644
--- a/media/codec2/vndk/C2Store.cpp
+++ b/media/codec2/vndk/C2Store.cpp
@@ -301,13 +301,21 @@ void C2PlatformAllocatorStoreImpl::setComponentStore(std::shared_ptr<C2Component
std::lock_guard<std::mutex> lock(_mComponentStoreReadLock);
_mComponentStore = store;
}
- std::shared_ptr<C2AllocatorIon> allocator;
+ std::shared_ptr<C2AllocatorIon> ionAllocator;
{
std::lock_guard<std::mutex> lock(gIonAllocatorMutex);
- allocator = gIonAllocator.lock();
+ ionAllocator = gIonAllocator.lock();
}
- if (allocator) {
- UseComponentStoreForIonAllocator(allocator, store);
+ if (ionAllocator) {
+ UseComponentStoreForIonAllocator(ionAllocator, store);
+ }
+ std::shared_ptr<C2DmaBufAllocator> dmaAllocator;
+ {
+ std::lock_guard<std::mutex> lock(gDmaBufAllocatorMutex);
+ dmaAllocator = gDmaBufAllocator.lock();
+ }
+ if (dmaAllocator) {
+ UseComponentStoreForDmaBufAllocator(dmaAllocator, store);
}
}
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index f07e66e1cc..90ff4a58bd 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -352,7 +352,8 @@ AAUDIO_API aaudio_result_t AAudioStream_waitForStateChange(AAudioStream* stream,
{
AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
- return audioStream->waitForStateChange(inputState, nextState, timeoutNanoseconds);
+ android::sp<AudioStream> spAudioStream(audioStream);
+ return spAudioStream->waitForStateChange(inputState, nextState, timeoutNanoseconds);
}
// ============================================================
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 73432af112..06f05b02ba 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -602,6 +602,7 @@ bool AudioStream::collidesWithCallback() const {
void AudioStream::setDuckAndMuteVolume(float duckAndMuteVolume) {
ALOGD("%s() to %f", __func__, duckAndMuteVolume);
+ std::lock_guard<std::mutex> lock(mStreamLock);
mDuckAndMuteVolume = duckAndMuteVolume;
doSetVolume(); // apply this change
}
diff --git a/media/libaaudio/tests/test_various.cpp b/media/libaaudio/tests/test_various.cpp
index cbf863fec7..b68fc7bd09 100644
--- a/media/libaaudio/tests/test_various.cpp
+++ b/media/libaaudio/tests/test_various.cpp
@@ -25,6 +25,7 @@
#include <gtest/gtest.h>
#include <unistd.h>
+#include <thread>
// Callback function that does nothing.
aaudio_data_callback_result_t NoopDataCallbackProc(
@@ -51,6 +52,7 @@ aaudio_data_callback_result_t NoopDataCallbackProc(
}
constexpr int64_t NANOS_PER_MILLISECOND = 1000 * 1000;
+constexpr int64_t MICROS_PER_MILLISECOND = 1000;
void checkReleaseThenClose(aaudio_performance_mode_t perfMode,
aaudio_sharing_mode_t sharingMode,
@@ -762,6 +764,58 @@ TEST(test_various, aaudio_callback_once_lowlat) {
checkCallbackOnce(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
}
+void waitForStateChangeToClosingorClosed(AAudioStream **stream, std::atomic<bool>* isReady)
+{
+ *isReady = true;
+ aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
+ EXPECT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(*stream,
+ AAUDIO_STREAM_STATE_OPEN, &state,
+ 10000 * NANOS_PER_MILLISECOND));
+ if ((state != AAUDIO_STREAM_STATE_CLOSING) && (state != AAUDIO_STREAM_STATE_CLOSED)){
+ FAIL() << "ERROR - State not closing or closed. Current state: " <<
+ AAudio_convertStreamStateToText(state);
+ }
+}
+
+void testWaitForStateChangeClose(aaudio_performance_mode_t perfMode) {
+ AAudioStreamBuilder *aaudioBuilder = nullptr;
+ AAudioStream *aaudioStream = nullptr;
+
+ ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+ AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
+ ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream));
+
+ // Verify Open State
+ aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
+ EXPECT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(aaudioStream,
+ AAUDIO_STREAM_STATE_UNKNOWN, &state,
+ 1000 * NANOS_PER_MILLISECOND));
+ EXPECT_EQ(AAUDIO_STREAM_STATE_OPEN, state);
+
+ std::atomic<bool> isWaitThreadReady{false};
+
+ // Spawn a new thread to wait for the state change
+ std::thread waitThread (waitForStateChangeToClosingorClosed, &aaudioStream,
+ &isWaitThreadReady);
+
+ // Wait for worker thread to be ready
+ while (!isWaitThreadReady) {
+ usleep(MICROS_PER_MILLISECOND);
+ }
+ // Sleep an additional millisecond to make sure waitForAudioThread is called
+ usleep(MICROS_PER_MILLISECOND);
+ EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream));
+ waitThread.join();
+}
+
+TEST(test_various, wait_for_state_change_close_none) {
+ testWaitForStateChangeClose(AAUDIO_PERFORMANCE_MODE_NONE);
+}
+
+TEST(test_various, wait_for_state_change_close_lowlat) {
+ testWaitForStateChangeClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+}
+
// ************************************************************
struct WakeUpCallbackData {
void wakeOther() {
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index c2bea66309..54d186e3b3 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -972,8 +972,16 @@ bool AudioTrack::pauseAndWait(const std::chrono::milliseconds& timeout)
{
using namespace std::chrono_literals;
+ // We use atomic access here for state variables - these are used as hints
+ // to ensure we have ramped down audio.
+ const int priorState = mProxy->getState();
+ const uint32_t priorPosition = mProxy->getPosition().unsignedValue();
+
pause();
+ // Only if we were previously active, do we wait to ramp down the audio.
+ if (priorState != CBLK_STATE_ACTIVE) return true;
+
AutoMutex lock(mLock);
// offload and direct tracks do not wait because pause volume ramp is handled by hardware.
if (isOffloadedOrDirect_l()) return true;
@@ -981,16 +989,25 @@ bool AudioTrack::pauseAndWait(const std::chrono::milliseconds& timeout)
// Wait for the track state to be anything besides pausing.
// This ensures that the volume has ramped down.
constexpr auto SLEEP_INTERVAL_MS = 10ms;
+ constexpr auto POSITION_TIMEOUT_MS = 40ms; // don't wait longer than this for position change.
auto begin = std::chrono::steady_clock::now();
while (true) {
- // wait for state to change
+ // Wait for state and position to change.
+ // After pause() the server state should be PAUSING, but that may immediately
+ // convert to PAUSED by prepareTracks before data is read into the mixer.
+ // Hence we check that the state is not PAUSING and that the server position
+ // has advanced to be a more reliable estimate that the volume ramp has completed.
const int state = mProxy->getState();
+ const uint32_t position = mProxy->getPosition().unsignedValue();
mLock.unlock(); // only local variables accessed until lock.
auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
std::chrono::steady_clock::now() - begin);
- if (state != CBLK_STATE_PAUSING) {
- ALOGV("%s: success state:%d after %lld ms", __func__, state, elapsed.count());
+ if (state != CBLK_STATE_PAUSING &&
+ (elapsed >= POSITION_TIMEOUT_MS || position != priorPosition)) {
+ ALOGV("%s: success state:%d, position:%u after %lld ms"
+ " (prior state:%d prior position:%u)",
+ __func__, state, position, elapsed.count(), priorState, priorPosition);
return true;
}
std::chrono::milliseconds remaining = timeout - elapsed;
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 1e64538d89..2c8d8de226 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -2195,7 +2195,10 @@ status_t ACodec::configureCodec(
}
if (!msg->findInt32("aac-max-output-channel_count", &maxOutputChannelCount)) {
- maxOutputChannelCount = -1;
+ // check non AAC-specific key
+ if (!msg->findInt32("max-output-channel-count", &maxOutputChannelCount)) {
+ maxOutputChannelCount = -1;
+ }
}
if (!msg->findInt32("aac-pcm-limiter-enable", &pcmLimiterEnable)) {
// value is unknown
diff --git a/media/libstagefright/include/media/stagefright/MediaCodecConstants.h b/media/libstagefright/include/media/stagefright/MediaCodecConstants.h
index 4237e8ccf7..c2c79dd396 100644
--- a/media/libstagefright/include/media/stagefright/MediaCodecConstants.h
+++ b/media/libstagefright/include/media/stagefright/MediaCodecConstants.h
@@ -789,6 +789,7 @@ constexpr char KEY_MAX_BIT_RATE[] = "max-bitrate";
constexpr char KEY_MAX_FPS_TO_ENCODER[] = "max-fps-to-encoder";
constexpr char KEY_MAX_HEIGHT[] = "max-height";
constexpr char KEY_MAX_INPUT_SIZE[] = "max-input-size";
+constexpr char KEY_MAX_OUTPUT_CHANNEL_COUNT[] = "max-output-channel-count";
constexpr char KEY_MAX_PTS_GAP_TO_ENCODER[] = "max-pts-gap-to-encoder";
constexpr char KEY_MAX_WIDTH[] = "max-width";
constexpr char KEY_MIME[] = "mime";
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index 0bd342ab01..83291f3f72 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -558,6 +558,7 @@ void ARTPConnection::onPollStreams() {
if (n != (ssize_t)buffer->size()) {
ALOGW("failed to send RTCP receiver report (%s).",
n >= 0 ? "connection gone" : strerror(errno));
+ ++it;
continue;
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index f3d2326601..866f1b5d15 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -549,6 +549,11 @@ status_t SwAudioOutputDescriptor::open(const audio_config_t *halConfig,
audio_output_flags_t halFlags = mFlags;
if ((mFlags & AUDIO_OUTPUT_FLAG_SPATIALIZER) != 0) {
halFlags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+ // If no mixer config is specified for a spatializer output, default to 5.1 for proper
+ // configuration of the final downmixer or spatializer
+ if (mixerConfig == nullptr) {
+ lMixerConfig.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+ }
}
ALOGV("opening output for device %s profile %p name %s",
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index e334532ebc..5a18762967 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1372,7 +1372,8 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevices(
}
if (mSpatializerOutput != nullptr
- && canBeSpatialized(attr, config, devices.toTypeAddrVector())) {
+ && canBeSpatializedInt(attr, config,
+ devices.toTypeAddrVector(), false /* allowCurrentOutputReconfig */)) {
return mSpatializerOutput->mIoHandle;
}
@@ -3011,9 +3012,10 @@ status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc,
}
}
}
- return mEffects.registerEffect(desc, io, session, id,
- (strategy == streamToStrategy(AUDIO_STREAM_MUSIC) ||
- strategy == PRODUCT_STRATEGY_NONE));
+ bool isMusicEffect = (session != AUDIO_SESSION_OUTPUT_STAGE)
+ && ((strategy == streamToStrategy(AUDIO_STREAM_MUSIC)
+ || strategy == PRODUCT_STRATEGY_NONE));
+ return mEffects.registerEffect(desc, io, session, id, isMusicEffect);
}
status_t AudioPolicyManager::unregisterEffect(int id)
@@ -4852,9 +4854,10 @@ bool AudioPolicyManager::isChannelMaskSpatialized(audio_channel_mask_t channels)
}
}
-bool AudioPolicyManager::canBeSpatialized(const audio_attributes_t *attr,
+bool AudioPolicyManager::canBeSpatializedInt(const audio_attributes_t *attr,
const audio_config_t *config,
- const AudioDeviceTypeAddrVector &devices) const
+ const AudioDeviceTypeAddrVector &devices,
+ bool allowCurrentOutputReconfig) const
{
// The caller can have the audio attributes criteria ignored by either passing a null ptr or
// the AUDIO_ATTRIBUTES_INITIALIZER value.
@@ -4890,7 +4893,8 @@ bool AudioPolicyManager::canBeSpatialized(const audio_attributes_t *attr,
if (!isChannelMaskSpatialized(config->channel_mask)) {
return false;
}
- if (mSpatializerOutput != nullptr && mSpatializerOutput->mProfile == profile) {
+ if (!allowCurrentOutputReconfig && mSpatializerOutput != nullptr
+ && mSpatializerOutput->mProfile == profile) {
if ((config->channel_mask & mSpatializerOutput->mMixerChannelMask)
!= config->channel_mask) {
return false;
@@ -4911,7 +4915,8 @@ void AudioPolicyManager::checkVirtualizerClientRoutes() {
audio_config_base_t clientConfig = client->config();
audio_config_t config = audio_config_initializer(&clientConfig);
if (desc != mSpatializerOutput
- && canBeSpatialized(&attr, &config, devicesTypeAddress)) {
+ && canBeSpatializedInt(&attr, &config,
+ devicesTypeAddress, false /* allowCurrentOutputReconfig */)) {
streamsToInvalidate.insert(client->stream());
}
}
@@ -4935,7 +4940,8 @@ status_t AudioPolicyManager::getSpatializerOutput(const audio_config_base_t *mix
config = audio_config_initializer(mixerConfig);
configPtr = &config;
}
- if (!canBeSpatialized(attr, configPtr, devicesTypeAddress)) {
+ if (!canBeSpatializedInt(
+ attr, configPtr, devicesTypeAddress)) {
ALOGW("%s provided attributes or mixer config cannot be spatialized", __func__);
return BAD_VALUE;
}
@@ -4958,6 +4964,7 @@ status_t AudioPolicyManager::getSpatializerOutput(const audio_config_base_t *mix
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (!desc->isDuplicated() && desc->mProfile == profile) {
+ ALOGV("%s found output %d for spatializer profile", __func__, desc->mIoHandle);
mSpatializerOutput = desc;
break;
}
@@ -4977,39 +4984,29 @@ status_t AudioPolicyManager::getSpatializerOutput(const audio_config_base_t *mix
};
DeviceVector savedDevices = mSpatializerOutput->devices();
+ ALOGV("%s reopening spatializer output to match channel mask %#x (current mask %#x)",
+ __func__, configPtr->channel_mask, mSpatializerOutput->mMixerChannelMask);
+
closeOutput(mSpatializerOutput->mIoHandle);
- mSpatializerOutput.clear();
+ //from now on mSpatializerOutput is null
- const sp<SwAudioOutputDescriptor> desc =
- new SwAudioOutputDescriptor(profile, mpClientInterface);
- status_t status = desc->open(nullptr, mixerConfig, devices,
- mEngine->getStreamTypeForAttributes(*attr),
- AUDIO_OUTPUT_FLAG_SPATIALIZER, output);
- if (status != NO_ERROR) {
- ALOGW("%s failed opening output: status %d, output %d", __func__, status, *output);
- if (*output != AUDIO_IO_HANDLE_NONE) {
- desc->close();
- }
+ sp<SwAudioOutputDescriptor> desc =
+ openOutputWithProfileAndDevice(profile, devices, mixerConfig);
+ if (desc == nullptr) {
// re open the spatializer output with previous channel mask
- status_t newStatus = desc->open(nullptr, &savedMixerConfig, savedDevices,
- mEngine->getStreamTypeForAttributes(*attr),
- AUDIO_OUTPUT_FLAG_SPATIALIZER, output);
- if (newStatus != NO_ERROR) {
- if (*output != AUDIO_IO_HANDLE_NONE) {
- desc->close();
- }
- ALOGE("%s failed to re-open mSpatializerOutput, status %d", __func__, newStatus);
+ desc = openOutputWithProfileAndDevice(profile, savedDevices, &savedMixerConfig);
+ if (desc == nullptr) {
+ ALOGE("%s failed to restore mSpatializerOutput with previous config", __func__);
} else {
mSpatializerOutput = desc;
- addOutput(*output, desc);
}
mPreviousOutputs = mOutputs;
mpClientInterface->onAudioPortListUpdate();
*output = AUDIO_IO_HANDLE_NONE;
- return status;
+ ALOGW("%s could not open spatializer output with requested config", __func__);
+ return BAD_VALUE;
}
mSpatializerOutput = desc;
- addOutput(*output, desc);
mPreviousOutputs = mOutputs;
mpClientInterface->onAudioPortListUpdate();
}
@@ -5682,6 +5679,9 @@ void AudioPolicyManager::closeOutput(audio_io_handle_t output)
removeOutput(output);
mPreviousOutputs = mOutputs;
+ if (closingOutput == mSpatializerOutput) {
+ mSpatializerOutput.clear();
+ }
// MSD patches may have been released to support a non-MSD direct output. Reset MSD patch if
// no direct outputs are open.
@@ -7222,7 +7222,8 @@ bool AudioPolicyManager::areAllActiveTracksRerouted(const sp<SwAudioOutputDescri
}
sp<SwAudioOutputDescriptor> AudioPolicyManager::openOutputWithProfileAndDevice(
- const sp<IOProfile>& profile, const DeviceVector& devices)
+ const sp<IOProfile>& profile, const DeviceVector& devices,
+ const audio_config_base_t *mixerConfig)
{
for (const auto& device : devices) {
// TODO: This should be checking if the profile supports the device combo.
@@ -7232,7 +7233,7 @@ sp<SwAudioOutputDescriptor> AudioPolicyManager::openOutputWithProfileAndDevice(
}
sp<SwAudioOutputDescriptor> desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- status_t status = desc->open(nullptr /* halConfig */, nullptr /* mixerConfig */, devices,
+ status_t status = desc->open(nullptr /* halConfig */, mixerConfig, devices,
AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
return nullptr;
@@ -7262,7 +7263,7 @@ sp<SwAudioOutputDescriptor> AudioPolicyManager::openOutputWithProfileAndDevice(
config.offload_info.channel_mask = config.channel_mask;
config.offload_info.format = config.format;
- status = desc->open(&config, nullptr /* mixerConfig */, devices,
+ status = desc->open(&config, mixerConfig, devices,
AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
return nullptr;
@@ -7270,6 +7271,7 @@ sp<SwAudioOutputDescriptor> AudioPolicyManager::openOutputWithProfileAndDevice(
}
addOutput(output, desc);
+
if (audio_is_remote_submix_device(deviceType) && address != "0") {
sp<AudioPolicyMix> policyMix;
if (mPolicyMixes.getAudioPolicyMix(deviceType, address, policyMix) == NO_ERROR) {
@@ -7280,9 +7282,13 @@ sp<SwAudioOutputDescriptor> AudioPolicyManager::openOutputWithProfileAndDevice(
address.string());
}
- } else if (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) && hasPrimaryOutput()) {
- // no duplicated output for direct outputs and
- // outputs used by dynamic policy mixes
+ } else if (hasPrimaryOutput() && profile->getModule()
+ != mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY)
+ && ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
+ // no duplicated output for:
+ // - direct outputs
+ // - outputs used by dynamic policy mixes
+ // - outputs opened on the primary HW module
audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE;
//TODO: configure audio effect output stage here
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 967aa10441..b1454fde47 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -358,7 +358,9 @@ public:
virtual bool canBeSpatialized(const audio_attributes_t *attr,
const audio_config_t *config,
- const AudioDeviceTypeAddrVector &devices) const;
+ const AudioDeviceTypeAddrVector &devices) const {
+ return canBeSpatializedInt(attr, config, devices);
+ }
virtual status_t getSpatializerOutput(const audio_config_base_t *config,
const audio_attributes_t *attr,
@@ -961,6 +963,30 @@ private:
const DeviceVector &devices,
audio_io_handle_t *output);
+ /**
+ * @brief Queries if some kind of spatialization will be performed if the audio playback
+ * context described by the provided arguments is present.
+ * The context is made of:
+ * - The audio attributes describing the playback use case.
+ * - The audio configuration describing the audio format, channels, sampling rate ...
+ * - The devices describing the sink audio device selected for playback.
+ * All arguments are optional and only the specified arguments are used to match against
+ * supported criteria. For instance, supplying no argument will tell if spatialization is
+ * supported or not in general.
+ * @param attr audio attributes describing the playback use case
+ * @param config audio configuration describing the audio format, channels, sample rate...
+ * @param devices the sink audio device selected for playback
+ * @param allowCurrentOutputReconfig if true, the result will be considering it is possible
+ * to close and reopen an existing spatializer output stream to match the requested
+ * criteria. If false, the criteria must be compatible with the opened sptializer
+ * output.
+ * @return true if spatialization is possible for this context, false otherwise.
+ */
+ virtual bool canBeSpatializedInt(const audio_attributes_t *attr,
+ const audio_config_t *config,
+ const AudioDeviceTypeAddrVector &devices,
+ bool allowCurrentOutputReconfig = true) const;
+
sp<IOProfile> getSpatializerOutputProfile(const audio_config_t *config,
const AudioDeviceTypeAddrVector &devices) const;
@@ -1056,8 +1082,20 @@ private:
bool areAllActiveTracksRerouted(const sp<SwAudioOutputDescriptor>& output);
- sp<SwAudioOutputDescriptor> openOutputWithProfileAndDevice(const sp<IOProfile>& profile,
- const DeviceVector& devices);
+ /**
+ * @brief Opens an output stream from the supplied IOProfile and route it to the
+ * supplied audio devices. If a mixer config is specified, it is forwarded to audio
+ * flinger. If not, a default config is derived from the output stream config.
+ * Also opens a duplicating output if needed and queries the audio HAL for supported
+ * audio profiles if the IOProfile is dynamic.
+ * @param[in] profile IOProfile to use as template
+ * @param[in] devices initial route to apply to this output stream
+ * @param[in] mixerConfig if not null, use this to configure the mixer
+ * @return an output descriptor for the newly opened stream or null in case of error.
+ */
+ sp<SwAudioOutputDescriptor> openOutputWithProfileAndDevice(
+ const sp<IOProfile>& profile, const DeviceVector& devices,
+ const audio_config_base_t *mixerConfig = nullptr);
};
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 102b376ded..12cd136db9 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -392,7 +392,8 @@ void AudioPolicyService::doOnCheckSpatializer()
audio_config_base_t config = mSpatializer->getAudioInConfig();
status_t status =
mAudioPolicyManager->getSpatializerOutput(&config, &attr, &newOutput);
-
+ ALOGV("%s currentOutput %d newOutput %d channel_mask %#x",
+ __func__, currentOutput, newOutput, config.channel_mask);
if (status == NO_ERROR && currentOutput == newOutput) {
return;
}
diff --git a/services/audiopolicy/service/Spatializer.cpp b/services/audiopolicy/service/Spatializer.cpp
index 0fdbe20239..6745005447 100644
--- a/services/audiopolicy/service/Spatializer.cpp
+++ b/services/audiopolicy/service/Spatializer.cpp
@@ -216,7 +216,9 @@ status_t Spatializer::loadEngineConfiguration(sp<EffectHalInterface> effect) {
if (status != NO_ERROR) {
return status;
}
- mSupportsHeadTracking = supportsHeadTracking[0];
+// Disable head tracking until head sensor activity is properly controlled.
+// mSupportsHeadTracking = supportsHeadTracking[0];
+ mSupportsHeadTracking = false;
status = getHalParameter<true>(effect, SPATIALIZER_PARAM_SUPPORTED_LEVELS, &mLevels);
if (status != NO_ERROR) {
@@ -227,12 +229,8 @@ status_t Spatializer::loadEngineConfiguration(sp<EffectHalInterface> effect) {
if (status != NO_ERROR) {
return status;
}
- status = getHalParameter<true>(effect, SPATIALIZER_PARAM_SUPPORTED_CHANNEL_MASKS,
+ return getHalParameter<true>(effect, SPATIALIZER_PARAM_SUPPORTED_CHANNEL_MASKS,
&mChannelMasks);
- if (status != NO_ERROR) {
- return status;
- }
- return NO_ERROR;
}
/** Gets the channel mask, sampling rate and format set for the spatializer input. */
@@ -727,7 +725,7 @@ void Spatializer::engineCallback(int32_t event, void *user, void *info) {
switch (event) {
case AudioEffect::EVENT_FRAMES_PROCESSED: {
int frames = info == nullptr ? 0 : *(int*)info;
- ALOGD("%s frames processed %d for me %p", __func__, frames, me);
+ // ALOGD("%s frames processed %d for me %p", __func__, frames, me);
me->postFramesProcessedMsg(frames);
} break;
default:
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-25 16:27:57 +0000