Merge "Refactored libcameraservice_depth_processor_fuzzer" into main
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.cpp b/media/libaaudio/src/core/AAudioStreamParameters.cpp
index 3e51575..67fc668 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.cpp
+++ b/media/libaaudio/src/core/AAudioStreamParameters.cpp
@@ -23,13 +23,6 @@
using namespace aaudio;
-// TODO These defines should be moved to a central place in audio.
-#define SAMPLES_PER_FRAME_MIN 1
-#define SAMPLES_PER_FRAME_MAX FCC_LIMIT
-#define SAMPLE_RATE_HZ_MIN 8000
-// HDMI supports up to 32 channels at 1536000 Hz.
-#define SAMPLE_RATE_HZ_MAX 1600000
-
void AAudioStreamParameters::copyFrom(const AAudioStreamParameters &other) {
mSamplesPerFrame = other.mSamplesPerFrame;
mSampleRate = other.mSampleRate;
@@ -73,8 +66,8 @@
}
aaudio_result_t AAudioStreamParameters::validate() const {
- if (mSamplesPerFrame != AAUDIO_UNSPECIFIED
- && (mSamplesPerFrame < SAMPLES_PER_FRAME_MIN || mSamplesPerFrame > SAMPLES_PER_FRAME_MAX)) {
+ if (mSamplesPerFrame != AAUDIO_UNSPECIFIED && (mSamplesPerFrame < CHANNEL_COUNT_MIN_AAUDIO ||
+ mSamplesPerFrame > CHANNEL_COUNT_MAX_AAUDIO)) {
ALOGD("channelCount out of range = %d", mSamplesPerFrame);
return AAUDIO_ERROR_OUT_OF_RANGE;
}
@@ -105,8 +98,8 @@
aaudio_result_t result = isFormatValid (mAudioFormat);
if (result != AAUDIO_OK) return result;
- if (mSampleRate != AAUDIO_UNSPECIFIED
- && (mSampleRate < SAMPLE_RATE_HZ_MIN || mSampleRate > SAMPLE_RATE_HZ_MAX)) {
+ if (mSampleRate != AAUDIO_UNSPECIFIED &&
+ (mSampleRate < SAMPLE_RATE_HZ_MIN_AAUDIO || mSampleRate > SAMPLE_RATE_HZ_MAX_IEC610937)) {
ALOGD("sampleRate out of range = %d", mSampleRate);
return AAUDIO_ERROR_INVALID_RATE;
}
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index ac4e2b3..4f32883 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -46,15 +46,6 @@
#define AAUDIO_MMAP_POLICY_DEFAULT AAUDIO_POLICY_NEVER
#define AAUDIO_MMAP_EXCLUSIVE_POLICY_DEFAULT AAUDIO_POLICY_NEVER
-// These values are for a pre-check before we ask the lower level service to open a stream.
-// So they are just outside the maximum conceivable range of value,
-// on the edge of being ridiculous.
-// TODO These defines should be moved to a central place in audio.
-#define SAMPLES_PER_FRAME_MIN 1
-#define SAMPLES_PER_FRAME_MAX FCC_LIMIT
-#define SAMPLE_RATE_HZ_MIN 8000
-// HDMI supports up to 32 channels at 1536000 Hz.
-#define SAMPLE_RATE_HZ_MAX 1600000
#define FRAMES_PER_DATA_CALLBACK_MIN 1
#define FRAMES_PER_DATA_CALLBACK_MAX (1024 * 1024)
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index e5323a6..d4b4b6c 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -278,10 +278,6 @@
"android.media.codec-aconfig-cc",
],
- include_dirs: [
- "system/libhidl/transport/token/1.0/utils/include",
- ],
-
export_include_dirs: [
"include",
],
diff --git a/media/libmedia/MediaCodecInfo.cpp b/media/libmedia/MediaCodecInfo.cpp
index c45c5c3..d5d1a09 100644
--- a/media/libmedia/MediaCodecInfo.cpp
+++ b/media/libmedia/MediaCodecInfo.cpp
@@ -18,8 +18,6 @@
#define LOG_TAG "MediaCodecInfo"
#include <utils/Log.h>
-#include <media/IOMX.h>
-
#include <media/MediaCodecInfo.h>
#include <media/stagefright/foundation/ADebug.h>
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 3ab32f0..736dd8a 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -1334,10 +1334,10 @@
// cause out-of-memory due to large input buffer size. And audio recording
// probably doesn't make sense in the scenario, since the slow-down factor
// is probably huge (eg. mSampleRate=48K, mCaptureFps=240, mFrameRate=1).
- const static int32_t SAMPLE_RATE_HZ_MAX = 192000;
+ const static int32_t kSampleRateHzMax = 192000;
sourceSampleRate =
(mSampleRate * mCaptureFps + mFrameRate / 2) / mFrameRate;
- if (sourceSampleRate < mSampleRate || sourceSampleRate > SAMPLE_RATE_HZ_MAX) {
+ if (sourceSampleRate < mSampleRate || sourceSampleRate > kSampleRateHzMax) {
ALOGE("source sample rate out of range! "
"(mSampleRate %d, mCaptureFps %.2f, mFrameRate %d",
mSampleRate, mCaptureFps, mFrameRate);
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 46c44a6..b073fa6 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -683,7 +683,7 @@
uint32_t mRightVolume; // previous volume on right channel
uint32_t mNewLeftVolume; // new volume on left channel
uint32_t mNewRightVolume; // new volume on right channel
- product_strategy_t mStrategy; // strategy for this effect chain
+ product_strategy_t mStrategy = PRODUCT_STRATEGY_NONE; // strategy for this effect chain
// mSuspendedEffects lists all effects currently suspended in the chain.
// Use effect type UUID timelow field as key. There is no real risk of identical
// timeLow fields among effect type UUIDs.
@@ -778,7 +778,7 @@
void checkSuspendOnEffectEnabled(const sp<IAfEffectBase>& effect __unused,
bool enabled __unused, bool threadLocked __unused) override {}
void resetVolume_l() override REQUIRES(audio_utils::EffectChain_Mutex) {}
- product_strategy_t strategy() const override { return static_cast<product_strategy_t>(0); }
+ product_strategy_t strategy() const override { return PRODUCT_STRATEGY_NONE; }
int32_t activeTrackCnt() const override { return 0; }
void onEffectEnable(const sp<IAfEffectBase>& effect __unused) override;
void onEffectDisable(const sp<IAfEffectBase>& effect __unused) override;
diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h
index 4643bd1..cf594c6 100644
--- a/services/audiopolicy/common/include/policy.h
+++ b/services/audiopolicy/common/include/policy.h
@@ -33,10 +33,6 @@
static const uint32_t SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY = 5000;
-// For mixed output and inputs, the policy will use max mixer sampling rates.
-// Do not limit sampling rate otherwise
-#define SAMPLE_RATE_HZ_MAX 192000
-
// Used when a client opens a capture stream, without specifying a desired sample rate.
#define SAMPLE_RATE_HZ_DEFAULT 48000
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index c502fc2..7002e63 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -282,6 +282,11 @@
const AudioProfileVector& getSupportedProfiles() { return mSupportedProfiles; }
+ /**
+ * @brief checks if all devices in device vector are attached to the HwModule or not
+ * @return true if all the devices in device vector are attached, otherwise false
+ */
+ bool areAllDevicesAttached() const;
// Return a string to describe the DeviceVector. The sensitive information will only be
// added to the string if `includeSensitiveInfo` is true.
std::string toString(bool includeSensitiveInfo = false) const;
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index 9f7b8fc..46a04de 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -541,4 +541,14 @@
return filteredDevices;
}
+bool DeviceVector::areAllDevicesAttached() const
+{
+ for (const auto &device : *this) {
+ if (!device->isAttached()) {
+ return false;
+ }
+ }
+ return true;
+}
+
} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp
index ce8178f..0ee84c1 100644
--- a/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp
@@ -20,6 +20,7 @@
#include "PolicyAudioPort.h"
#include "HwModule.h"
#include <policy.h>
+#include <system/audio.h>
#ifndef ARRAY_SIZE
#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 8561a9d..dffe259 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -374,6 +374,7 @@
checkLeBroadcastRoutes(wasLeUnicastActive, nullptr, 0);
mpClientInterface->onAudioPortListUpdate();
+ ALOGV("%s() completed for device: %s", __func__, device->toString().c_str());
return NO_ERROR;
} // end if is output device
@@ -389,6 +390,8 @@
return INVALID_OPERATION;
}
+ ALOGV("%s() connecting device %s", __func__, device->toString().c_str());
+
if (mAvailableInputDevices.add(device) < 0) {
return NO_MEMORY;
}
@@ -461,6 +464,7 @@
}
mpClientInterface->onAudioPortListUpdate();
+ ALOGV("%s() completed for device: %s", __func__, device->toString().c_str());
return NO_ERROR;
} // end if is input device
@@ -1516,7 +1520,7 @@
(config->channel_mask == desc->getChannelMask()) &&
(session == desc->mDirectClientSession)) {
desc->mDirectOpenCount++;
- ALOGV("%s reusing direct output %d for session %d", __func__,
+ ALOGI("%s reusing direct output %d for session %d", __func__,
mOutputs.keyAt(i), session);
*output = mOutputs.keyAt(i);
return NO_ERROR;
@@ -1526,17 +1530,23 @@
if (!profile->canOpenNewIo()) {
if (!com::android::media::audioserver::direct_track_reprioritization()) {
+ ALOGW("%s profile %s can't open new output maxOpenCount reached", __func__,
+ profile->getName().c_str());
return NAME_NOT_FOUND;
} else if ((profile->getFlags() & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) != 0) {
// MMAP gracefully handles lack of an exclusive track resource by mixing
// above the audio framework. For AAudio to know that the limit is reached,
// return an error.
+ ALOGW("%s profile %s can't open new mmap output maxOpenCount reached", __func__,
+ profile->getName().c_str());
return NAME_NOT_FOUND;
} else {
// Close outputs on this profile, if available, to free resources for this request
for (int i = 0; i < mOutputs.size() && !profile->canOpenNewIo(); i++) {
const auto desc = mOutputs.valueAt(i);
if (desc->mProfile == profile) {
+ ALOGV("%s closeOutput %d to prioritize session %d on profile %s", __func__,
+ desc->mIoHandle, session, profile->getName().c_str());
closeOutput(desc->mIoHandle);
}
}
@@ -1545,6 +1555,8 @@
// Unable to close streams to find free resources for this request
if (!profile->canOpenNewIo()) {
+ ALOGW("%s profile %s can't open new output maxOpenCount reached", __func__,
+ profile->getName().c_str());
return NAME_NOT_FOUND;
}
@@ -3276,8 +3288,8 @@
ALOGW("%s: no group for stream %s, bailing out", __func__, toString(stream).c_str());
return NO_ERROR;
}
- ALOGV("%s: stream %s attributes=%s", __func__,
- toString(stream).c_str(), toString(attributes).c_str());
+ ALOGV("%s: stream %s attributes=%s, index %d , device 0x%X", __func__,
+ toString(stream).c_str(), toString(attributes).c_str(), index, device);
return setVolumeIndexForAttributes(attributes, index, device);
}
@@ -3455,8 +3467,8 @@
bool hasVoice = hasVoiceStream(volumeCurves.getStreamTypes());
if (((index < volumeCurves.getVolumeIndexMin()) && !(hasVoice && index == 0)) ||
(index > volumeCurves.getVolumeIndexMax())) {
- ALOGD("%s: wrong index %d min=%d max=%d", __FUNCTION__, index,
- volumeCurves.getVolumeIndexMin(), volumeCurves.getVolumeIndexMax());
+ ALOGE("%s: wrong index %d min=%d max=%d, device 0x%X", __FUNCTION__, index,
+ volumeCurves.getVolumeIndexMin(), volumeCurves.getVolumeIndexMax(), device);
return BAD_VALUE;
}
if (!audio_is_output_device(device)) {
@@ -6386,6 +6398,14 @@
if (!mConfig->getOutputDevices().contains(supportedDevice)) {
continue;
}
+
+ if (outProfile->isMmap() && !outProfile->hasDynamicAudioProfile()
+ && availProfileDevices.areAllDevicesAttached()) {
+ ALOGV("%s skip opening output for mmap profile %s", __func__,
+ outProfile->getTagName().c_str());
+ continue;
+ }
+
sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
mpClientInterface);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
@@ -6445,6 +6465,14 @@
__func__, inProfile->getTagName().c_str());
continue;
}
+
+ if (inProfile->isMmap() && !inProfile->hasDynamicAudioProfile()
+ && availProfileDevices.areAllDevicesAttached()) {
+ ALOGV("%s skip opening input for mmap profile %s", __func__,
+ inProfile->getTagName().c_str());
+ continue;
+ }
+
sp<AudioInputDescriptor> inputDesc =
new AudioInputDescriptor(inProfile, mpClientInterface);
@@ -6452,12 +6480,12 @@
status_t status = inputDesc->open(nullptr,
availProfileDevices.itemAt(0),
AUDIO_SOURCE_MIC,
- AUDIO_INPUT_FLAG_NONE,
+ (audio_input_flags_t) inProfile->getFlags(),
&input);
if (status != NO_ERROR) {
- ALOGW("Cannot open input stream for device %s on hw module %s",
- availProfileDevices.toString().c_str(),
- hwModule->getName());
+ ALOGW("%s: Cannot open input stream for device %s for profile %s on hw module %s",
+ __func__, availProfileDevices.toString().c_str(),
+ inProfile->getTagName().c_str(), hwModule->getName());
continue;
}
for (const auto &device : availProfileDevices) {
@@ -6565,8 +6593,8 @@
sp<IOProfile> profile = hwModule->getOutputProfiles()[j];
if (profile->supportsDevice(device)) {
profiles.add(profile);
- ALOGV("checkOutputsForDevice(): adding profile %zu from module %s",
- j, hwModule->getName());
+ ALOGV("%s(): adding profile %s from module %s",
+ __func__, profile->getTagName().c_str(), hwModule->getName());
}
}
}
@@ -6599,7 +6627,11 @@
if (j != outputs.size()) {
continue;
}
-
+ if (profile->isMmap() && !profile->hasDynamicAudioProfile()) {
+ ALOGV("%s skip opening output for mmap profile %s",
+ __func__, profile->getTagName().c_str());
+ continue;
+ }
if (!profile->canOpenNewIo()) {
ALOGW("Max Output number %u already opened for this profile %s",
profile->maxOpenCount, profile->getTagName().c_str());
@@ -6660,9 +6692,8 @@
if (!profile->supportsDevice(device)) {
continue;
}
- ALOGV("checkOutputsForDevice(): "
- "clearing direct output profile %zu on module %s",
- j, hwModule->getName());
+ ALOGV("%s(): clearing direct output profile %s on module %s",
+ __func__, profile->getTagName().c_str(), hwModule->getName());
profile->clearAudioProfiles();
if (!profile->hasDynamicAudioProfile()) {
continue;
@@ -6717,8 +6748,8 @@
if (profile->supportsDevice(device)) {
profiles.add(profile);
- ALOGV("checkInputsForDevice(): adding profile %zu from module %s",
- profile_index, hwModule->getName());
+ ALOGV("%s : adding profile %s from module %s", __func__,
+ profile->getTagName().c_str(), hwModule->getName());
}
}
}
@@ -6750,15 +6781,22 @@
continue;
}
+ if (profile->isMmap() && !profile->hasDynamicAudioProfile()) {
+ ALOGV("%s skip opening input for mmap profile %s",
+ __func__, profile->getTagName().c_str());
+ continue;
+ }
if (!profile->canOpenNewIo()) {
- ALOGW("Max Input number %u already opened for this profile %s",
- profile->maxOpenCount, profile->getTagName().c_str());
+ ALOGW("%s Max Input number %u already opened for this profile %s",
+ __func__, profile->maxOpenCount, profile->getTagName().c_str());
continue;
}
desc = new AudioInputDescriptor(profile, mpClientInterface);
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
- status = desc->open(nullptr, device, AUDIO_SOURCE_MIC, AUDIO_INPUT_FLAG_NONE, &input);
+ ALOGV("%s opening input for profile %s", __func__, profile->getTagName().c_str());
+ status = desc->open(nullptr, device, AUDIO_SOURCE_MIC,
+ (audio_input_flags_t) profile->getFlags(), &input);
if (status == NO_ERROR) {
const String8& address = String8(device->address().c_str());
@@ -6769,7 +6807,8 @@
}
updateAudioProfiles(device, input, profile);
if (!profile->hasValidAudioProfile()) {
- ALOGW("checkInputsForDevice() direct input missing param");
+ ALOGW("%s direct input missing param for profile %s", __func__,
+ profile->getTagName().c_str());
desc->close();
input = AUDIO_IO_HANDLE_NONE;
}
@@ -6780,18 +6819,20 @@
} // endif input != 0
if (input == AUDIO_IO_HANDLE_NONE) {
- ALOGW("%s could not open input for device %s", __func__,
- device->toString().c_str());
+ ALOGW("%s could not open input for device %s on profile %s", __func__,
+ device->toString().c_str(), profile->getTagName().c_str());
profiles.removeAt(profile_index);
profile_index--;
} else {
if (audio_device_is_digital(device->type())) {
device->importAudioPortAndPickAudioProfile(profile);
}
- ALOGV("checkInputsForDevice(): adding input %d", input);
+ ALOGV("%s: adding input %d for profile %s", __func__,
+ input, profile->getTagName().c_str());
if (checkCloseInput(desc)) {
- ALOGV("%s closing input %d", __func__, input);
+ ALOGV("%s: closing input %d for profile %s", __func__,
+ input, profile->getTagName().c_str());
closeInput(input);
}
}
@@ -6810,8 +6851,8 @@
profile_index++) {
sp<IOProfile> profile = hwModule->getInputProfiles()[profile_index];
if (profile->supportsDevice(device)) {
- ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %s",
- profile_index, hwModule->getName());
+ ALOGV("%s: clearing direct input profile %s on module %s", __func__,
+ profile->getTagName().c_str(), hwModule->getName());
profile->clearAudioProfiles();
}
}