Merge "[clang-tidy] Disable bugprone-narrowing-conversions for mediametrics"
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 9cabd8b..200e92d 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -182,6 +182,7 @@
// This is set by AudioTrack.setBufferSizeInFrames().
// A write will not fill the buffer above this limit.
volatile uint32_t mBufferSizeInFrames; // effective size of the buffer
+ volatile uint32_t mStartThresholdInFrames; // min frames in buffer to start streaming
public:
@@ -216,6 +217,8 @@
};
size_t frameCount() const { return mFrameCount; }
+ uint32_t getStartThresholdInFrames() const;
+ uint32_t setStartThresholdInFrames(uint32_t startThresholdInFrames);
protected:
// These refer to shared memory, and are virtual addresses with respect to the current process.
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 19d1d1a..e37cc12 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -1249,6 +1249,46 @@
return finalBufferSize;
}
+ssize_t AudioTrack::getStartThresholdInFrames() const
+{
+ AutoMutex lock(mLock);
+ if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
+ return NO_INIT;
+ }
+ return (ssize_t) mProxy->getStartThresholdInFrames();
+}
+
+ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
+{
+ if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
+ // contractually we could simply return the current threshold in frames
+ // to indicate the request was ignored, but we return an error here.
+ return BAD_VALUE;
+ }
+ AutoMutex lock(mLock);
+ // We do not permit calling setStartThresholdInFrames() between the AudioTrack
+ // default ctor AudioTrack() and set(...) but rather fail such an attempt.
+ // (To do so would require a cached mOrigStartThresholdInFrames and we may
+ // not have proper validation for the actual set value).
+ if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
+ return NO_INIT;
+ }
+ const uint32_t original = mProxy->getStartThresholdInFrames();
+ const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
+ if (original != final) {
+ android::mediametrics::LogItem(mMetricsId)
+ .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
+ .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
+ .record();
+ if (original > final) {
+ // restart track if it was disabled by audioflinger due to previous underrun
+ // and we reduced the number of frames for the threshold.
+ restartIfDisabled();
+ }
+ }
+ return final;
+}
+
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
@@ -2562,6 +2602,10 @@
staticPosition = mStaticProxy->getPosition().unsignedValue();
}
+ // save the old startThreshold and framecount
+ const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
+ const uint32_t originalFrameCount = mProxy->frameCount();
+
// See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
// causes a lot of churn on the service side, and it can reject starting
// playback of a previously created track. May also apply to other cases.
@@ -2616,6 +2660,18 @@
return mAudioTrack->applyVolumeShaper(shaper.mConfiguration, operationToEnd);
});
+ // restore the original start threshold if different than frameCount.
+ if (originalStartThresholdInFrames != originalFrameCount) {
+ // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
+ // and does not trigger a restart.
+ // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
+ // Any start would be triggered on the mState == ACTIVE check below.
+ const uint32_t currentThreshold =
+ mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
+ ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
+ "%s(%d) startThresholdInFrames changing from %u to %u",
+ __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
+ }
if (mState == STATE_ACTIVE) {
result = mAudioTrack->start();
}
diff --git a/media/libaudioclient/AudioTrackShared.cpp b/media/libaudioclient/AudioTrackShared.cpp
index f1f8f9c..35719be 100644
--- a/media/libaudioclient/AudioTrackShared.cpp
+++ b/media/libaudioclient/AudioTrackShared.cpp
@@ -17,6 +17,7 @@
#define LOG_TAG "AudioTrackShared"
//#define LOG_NDEBUG 0
+#include <atomic>
#include <android-base/macros.h>
#include <private/media/AudioTrackShared.h>
#include <utils/Log.h>
@@ -33,6 +34,21 @@
return sizeof(T) > sizeof(size_t) && x > (T) SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x;
}
+// compile-time safe atomics. TODO: update all methods to use it
+template <typename T>
+T android_atomic_load(const volatile T* addr) {
+ static_assert(sizeof(T) == sizeof(std::atomic<T>)); // no extra sync data required.
+ static_assert(std::atomic<T>::is_always_lock_free); // no hash lock somewhere.
+ return atomic_load((std::atomic<T>*)addr); // memory_order_seq_cst
+}
+
+template <typename T>
+void android_atomic_store(const volatile T* addr, T value) {
+ static_assert(sizeof(T) == sizeof(std::atomic<T>)); // no extra sync data required.
+ static_assert(std::atomic<T>::is_always_lock_free); // no hash lock somewhere.
+ atomic_store((std::atomic<T>*)addr, value); // memory_order_seq_cst
+}
+
// incrementSequence is used to determine the next sequence value
// for the loop and position sequence counters. It should return
// a value between "other" + 1 and "other" + INT32_MAX, the choice of
@@ -51,6 +67,7 @@
: mServer(0), mFutex(0), mMinimum(0)
, mVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY), mSampleRate(0), mSendLevel(0)
, mBufferSizeInFrames(0)
+ , mStartThresholdInFrames(0) // filled in by the server.
, mFlags(0)
{
memset(&u, 0, sizeof(u));
@@ -66,6 +83,26 @@
{
}
+uint32_t Proxy::getStartThresholdInFrames() const
+{
+ const uint32_t startThresholdInFrames =
+ android_atomic_load(&mCblk->mStartThresholdInFrames);
+ if (startThresholdInFrames == 0 || startThresholdInFrames > mFrameCount) {
+ ALOGD("%s: startThresholdInFrames %u not between 1 and frameCount %zu, "
+ "setting to frameCount",
+ __func__, startThresholdInFrames, mFrameCount);
+ return mFrameCount;
+ }
+ return startThresholdInFrames;
+}
+
+uint32_t Proxy::setStartThresholdInFrames(uint32_t startThresholdInFrames)
+{
+ const uint32_t actual = std::min((size_t)startThresholdInFrames, frameCount());
+ android_atomic_store(&mCblk->mStartThresholdInFrames, actual);
+ return actual;
+}
+
// ---------------------------------------------------------------------------
ClientProxy::ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
@@ -663,6 +700,7 @@
, mTimestampMutator(&cblk->mExtendedTimestampQueue)
{
cblk->mBufferSizeInFrames = frameCount;
+ cblk->mStartThresholdInFrames = frameCount;
}
__attribute__((no_sanitize("integer")))
@@ -900,11 +938,8 @@
}
audio_track_cblk_t* cblk = mCblk;
- int32_t flush = cblk->u.mStreaming.mFlush;
- if (flush != mFlush) {
- // FIXME should return an accurate value, but over-estimate is better than under-estimate
- return mFrameCount;
- }
+ flushBufferIfNeeded();
+
const int32_t rear = getRear();
ssize_t filled = audio_utils::safe_sub_overflow(rear, cblk->u.mStreaming.mFront);
// pipe should not already be overfull
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index fac4c83..4afa9c9 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -429,6 +429,19 @@
*/
ssize_t setBufferSizeInFrames(size_t size);
+ /* Returns the start threshold on the buffer for audio streaming
+ * or a negative value if the AudioTrack is not initialized.
+ */
+ ssize_t getStartThresholdInFrames() const;
+
+ /* Sets the start threshold in frames on the buffer for audio streaming.
+ *
+ * May be clamped internally. Returns the actual value set, or a negative
+ * value if the AudioTrack is not initialized or if the input
+ * is zero or greater than INT_MAX.
+ */
+ ssize_t setStartThresholdInFrames(size_t startThresholdInFrames);
+
/* Return the static buffer specified in constructor or set(), or 0 for streaming mode */
sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index da16477..0d5fe59 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -246,6 +246,10 @@
return status;
}
CoreUtils::AudioInputFlags hidlFlags;
+#if MAJOR_VERSION <= 5
+ // Some flags were specific to framework and must not leak to the HAL.
+ flags = static_cast<audio_input_flags_t>(flags & ~AUDIO_INPUT_FLAG_DIRECT);
+#endif
if (status_t status = CoreUtils::audioInputFlagsFromHal(flags, &hidlFlags); status != OK) {
return status;
}
@@ -278,10 +282,6 @@
sinkMetadata.tracks[0].destination.device(std::move(hidlOutputDevice));
}
#endif
-#if MAJOR_VERSION <= 5
- // Some flags were specific to framework and must not leak to the HAL.
- flags = static_cast<audio_input_flags_t>(flags & ~AUDIO_INPUT_FLAG_DIRECT);
-#endif
Return<void> ret = mDevice->openInputStream(
handle, hidlDevice, hidlConfig, hidlFlags, sinkMetadata,
[&](Result r, const sp<IStreamIn>& result, const AudioConfig& suggestedConfig) {
diff --git a/media/libeffects/preprocessing/Android.bp b/media/libeffects/preprocessing/Android.bp
index 87ed8b6..c6e036a 100644
--- a/media/libeffects/preprocessing/Android.bp
+++ b/media/libeffects/preprocessing/Android.bp
@@ -18,15 +18,10 @@
],
}
-cc_library {
- name: "libaudiopreprocessing",
+cc_defaults {
+ name: "libaudiopreprocessing-defaults",
vendor: true,
- relative_install_path: "soundfx",
host_supported: true,
- srcs: ["PreProcessing.cpp"],
- local_include_dirs: [
- ".",
- ],
cflags: [
"-Wall",
"-Werror",
@@ -46,7 +41,6 @@
header_libs: [
"libaudioeffects",
"libhardware_headers",
- "libwebrtc_absl_headers",
],
target: {
darwin: {
@@ -54,3 +48,13 @@
},
},
}
+
+cc_library {
+ name: "libaudiopreprocessing",
+ defaults: ["libaudiopreprocessing-defaults"],
+ relative_install_path: "soundfx",
+ srcs: ["PreProcessing.cpp"],
+ header_libs: [
+ "libwebrtc_absl_headers",
+ ],
+}
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index 3b0b6d6..19a8b2f 100644
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -105,9 +105,8 @@
webrtc::AudioProcessing* apm; // handle on webRTC audio processing module (APM)
// Audio Processing module builder
webrtc::AudioProcessingBuilder ap_builder;
- size_t apmFrameCount; // buffer size for webRTC process (10 ms)
- uint32_t apmSamplingRate; // webRTC APM sampling rate (8/16 or 32 kHz)
- size_t frameCount; // buffer size before input resampler ( <=> apmFrameCount)
+ // frameCount represents the size of the buffers used for processing, and must represent 10ms.
+ size_t frameCount;
uint32_t samplingRate; // sampling rate at effect process interface
uint32_t inChannelCount; // input channel count
uint32_t outChannelCount; // output channel count
@@ -119,21 +118,12 @@
webrtc::AudioProcessing::Config config;
webrtc::StreamConfig inputConfig; // input stream configuration
webrtc::StreamConfig outputConfig; // output stream configuration
- int16_t* inBuf; // input buffer used when resampling
- size_t inBufSize; // input buffer size in frames
- size_t framesIn; // number of frames in input buffer
- int16_t* outBuf; // output buffer used when resampling
- size_t outBufSize; // output buffer size in frames
- size_t framesOut; // number of frames in output buffer
uint32_t revChannelCount; // number of channels on reverse stream
uint32_t revEnabledMsk; // bit field containing IDs of enabled pre processors
// with reverse channel
uint32_t revProcessedMsk; // bit field containing IDs of pre processors with reverse
// channel already processed in current round
webrtc::StreamConfig revConfig; // reverse stream configuration.
- int16_t* revBuf; // reverse channel input buffer
- size_t revBufSize; // reverse channel input buffer size
- size_t framesRev; // number of frames in reverse channel input buffer
};
#ifdef DUAL_MIC_TEST
@@ -862,9 +852,7 @@
ALOGW("Session_CreateEffect could not get apm engine");
goto error;
}
- session->apmSamplingRate = kPreprocDefaultSr;
- session->apmFrameCount = (kPreprocDefaultSr) / 100;
- session->frameCount = session->apmFrameCount;
+ session->frameCount = kPreprocDefaultSr / 100;
session->samplingRate = kPreprocDefaultSr;
session->inChannelCount = kPreProcDefaultCnl;
session->outChannelCount = kPreProcDefaultCnl;
@@ -879,12 +867,6 @@
session->processedMsk = 0;
session->revEnabledMsk = 0;
session->revProcessedMsk = 0;
- session->inBuf = NULL;
- session->inBufSize = 0;
- session->outBuf = NULL;
- session->outBufSize = 0;
- session->revBuf = NULL;
- session->revBufSize = 0;
}
status = Effect_Create(&session->effects[procId], session, interface);
if (status < 0) {
@@ -908,13 +890,6 @@
if (session->createdMsk == 0) {
delete session->apm;
session->apm = NULL;
- delete session->inBuf;
- session->inBuf = NULL;
- free(session->outBuf);
- session->outBuf = NULL;
- delete session->revBuf;
- session->revBuf = NULL;
-
session->id = 0;
}
@@ -934,24 +909,8 @@
ALOGV("Session_SetConfig sr %d cnl %08x", config->inputCfg.samplingRate,
config->inputCfg.channels);
- // AEC implementation is limited to 16kHz
- if (config->inputCfg.samplingRate >= 32000 && !(session->createdMsk & (1 << PREPROC_AEC))) {
- session->apmSamplingRate = 32000;
- } else if (config->inputCfg.samplingRate >= 16000) {
- session->apmSamplingRate = 16000;
- } else if (config->inputCfg.samplingRate >= 8000) {
- session->apmSamplingRate = 8000;
- }
-
-
session->samplingRate = config->inputCfg.samplingRate;
- session->apmFrameCount = session->apmSamplingRate / 100;
- if (session->samplingRate == session->apmSamplingRate) {
- session->frameCount = session->apmFrameCount;
- } else {
- session->frameCount =
- (session->apmFrameCount * session->samplingRate) / session->apmSamplingRate;
- }
+ session->frameCount = session->samplingRate / 100;
session->inChannelCount = inCnl;
session->outChannelCount = outCnl;
session->inputConfig.set_sample_rate_hz(session->samplingRate);
@@ -963,13 +922,6 @@
session->revConfig.set_sample_rate_hz(session->samplingRate);
session->revConfig.set_num_channels(inCnl);
- // force process buffer reallocation
- session->inBufSize = 0;
- session->outBufSize = 0;
- session->framesIn = 0;
- session->framesOut = 0;
-
-
session->state = PREPROC_SESSION_STATE_CONFIG;
return 0;
}
@@ -1004,9 +956,6 @@
}
uint32_t inCnl = audio_channel_count_from_out_mask(config->inputCfg.channels);
session->revChannelCount = inCnl;
- // force process buffer reallocation
- session->revBufSize = 0;
- session->framesRev = 0;
return 0;
}
@@ -1023,12 +972,8 @@
void Session_SetProcEnabled(preproc_session_t* session, uint32_t procId, bool enabled) {
if (enabled) {
- if (session->enabledMsk == 0) {
- session->framesIn = 0;
- }
session->enabledMsk |= (1 << procId);
if (HasReverseStream(procId)) {
- session->framesRev = 0;
session->revEnabledMsk |= (1 << procId);
}
} else {
@@ -1117,43 +1062,24 @@
return -EINVAL;
}
+ if (inBuffer->frameCount != outBuffer->frameCount) {
+ ALOGW("inBuffer->frameCount %zu is not equal to outBuffer->frameCount %zu",
+ inBuffer->frameCount, outBuffer->frameCount);
+ return -EINVAL;
+ }
+
+ if (inBuffer->frameCount != session->frameCount) {
+ ALOGW("inBuffer->frameCount %zu != %zu representing 10ms at sampling rate %d",
+ inBuffer->frameCount, session->frameCount, session->samplingRate);
+ return -EINVAL;
+ }
+
session->processedMsk |= (1 << effect->procId);
// ALOGV("PreProcessingFx_Process In %d frames enabledMsk %08x processedMsk %08x",
// inBuffer->frameCount, session->enabledMsk, session->processedMsk);
-
if ((session->processedMsk & session->enabledMsk) == session->enabledMsk) {
effect->session->processedMsk = 0;
- size_t framesRq = outBuffer->frameCount;
- size_t framesWr = 0;
- if (session->framesOut) {
- size_t fr = session->framesOut;
- if (outBuffer->frameCount < fr) {
- fr = outBuffer->frameCount;
- }
- memcpy(outBuffer->s16, session->outBuf,
- fr * session->outChannelCount * sizeof(int16_t));
- memmove(session->outBuf, session->outBuf + fr * session->outChannelCount,
- (session->framesOut - fr) * session->outChannelCount * sizeof(int16_t));
- session->framesOut -= fr;
- framesWr += fr;
- }
- outBuffer->frameCount = framesWr;
- if (framesWr == framesRq) {
- inBuffer->frameCount = 0;
- return 0;
- }
-
- size_t fr = session->frameCount - session->framesIn;
- if (inBuffer->frameCount < fr) {
- fr = inBuffer->frameCount;
- }
- session->framesIn += fr;
- inBuffer->frameCount = fr;
- if (session->framesIn < session->frameCount) {
- return 0;
- }
- session->framesIn = 0;
if (int status = effect->session->apm->ProcessStream(
(const int16_t* const)inBuffer->s16,
(const webrtc::StreamConfig)effect->session->inputConfig,
@@ -1163,34 +1089,6 @@
ALOGE("Process Stream failed with error %d\n", status);
return status;
}
- outBuffer->frameCount = inBuffer->frameCount;
-
- if (session->outBufSize < session->framesOut + session->frameCount) {
- int16_t* buf;
- session->outBufSize = session->framesOut + session->frameCount;
- buf = (int16_t*)realloc(
- session->outBuf,
- session->outBufSize * session->outChannelCount * sizeof(int16_t));
- if (buf == NULL) {
- session->framesOut = 0;
- free(session->outBuf);
- session->outBuf = NULL;
- return -ENOMEM;
- }
- session->outBuf = buf;
- }
-
- fr = session->framesOut;
- if (framesRq - framesWr < fr) {
- fr = framesRq - framesWr;
- }
- memcpy(outBuffer->s16 + framesWr * session->outChannelCount, session->outBuf,
- fr * session->outChannelCount * sizeof(int16_t));
- memmove(session->outBuf, session->outBuf + fr * session->outChannelCount,
- (session->framesOut - fr) * session->outChannelCount * sizeof(int16_t));
- session->framesOut -= fr;
- outBuffer->frameCount += fr;
-
return 0;
} else {
return -ENODATA;
@@ -1565,6 +1463,18 @@
return -EINVAL;
}
+ if (inBuffer->frameCount != outBuffer->frameCount) {
+ ALOGW("inBuffer->frameCount %zu is not equal to outBuffer->frameCount %zu",
+ inBuffer->frameCount, outBuffer->frameCount);
+ return -EINVAL;
+ }
+
+ if (inBuffer->frameCount != session->frameCount) {
+ ALOGW("inBuffer->frameCount %zu != %zu representing 10ms at sampling rate %d",
+ inBuffer->frameCount, session->frameCount, session->samplingRate);
+ return -EINVAL;
+ }
+
session->revProcessedMsk |= (1 << effect->procId);
// ALOGV("PreProcessingFx_ProcessReverse In %d frames revEnabledMsk %08x revProcessedMsk
@@ -1573,16 +1483,6 @@
if ((session->revProcessedMsk & session->revEnabledMsk) == session->revEnabledMsk) {
effect->session->revProcessedMsk = 0;
- size_t fr = session->frameCount - session->framesRev;
- if (inBuffer->frameCount < fr) {
- fr = inBuffer->frameCount;
- }
- session->framesRev += fr;
- inBuffer->frameCount = fr;
- if (session->framesRev < session->frameCount) {
- return 0;
- }
- session->framesRev = 0;
if (int status = effect->session->apm->ProcessReverseStream(
(const int16_t* const)inBuffer->s16,
(const webrtc::StreamConfig)effect->session->revConfig,
diff --git a/media/libeffects/preprocessing/README.md b/media/libeffects/preprocessing/README.md
new file mode 100644
index 0000000..af46376
--- /dev/null
+++ b/media/libeffects/preprocessing/README.md
@@ -0,0 +1,7 @@
+# Preprocessing effects
+
+## Limitations
+- Preprocessing effects currently work on 10ms worth of data and do not support
+ arbitrary frame counts. This limiation comes from the underlying effects in
+ webrtc modules
+- There is currently no api to communicate this requirement
diff --git a/media/libeffects/preprocessing/benchmarks/Android.bp b/media/libeffects/preprocessing/benchmarks/Android.bp
index c1b2295..fbbcab4 100644
--- a/media/libeffects/preprocessing/benchmarks/Android.bp
+++ b/media/libeffects/preprocessing/benchmarks/Android.bp
@@ -11,27 +11,10 @@
cc_benchmark {
name: "preprocessing_benchmark",
- vendor: true,
+ defaults: ["libaudiopreprocessing-defaults"],
srcs: ["preprocessing_benchmark.cpp"],
- shared_libs: [
- "libaudioutils",
- "liblog",
- "libutils",
- ],
static_libs: [
"libaudiopreprocessing",
- "webrtc_audio_processing",
- ],
- cflags: [
- "-DWEBRTC_POSIX",
- "-fvisibility=default",
- "-Wall",
- "-Werror",
- "-Wextra",
- ],
- header_libs: [
- "libaudioeffects",
- "libhardware_headers",
- "libwebrtc_absl_headers",
+ "libaudioutils",
],
}
diff --git a/media/libeffects/preprocessing/tests/Android.bp b/media/libeffects/preprocessing/tests/Android.bp
index 18c6c98..d80b135 100644
--- a/media/libeffects/preprocessing/tests/Android.bp
+++ b/media/libeffects/preprocessing/tests/Android.bp
@@ -12,9 +12,8 @@
cc_test {
name: "EffectPreprocessingTest",
- vendor: true,
+ defaults: ["libaudiopreprocessing-defaults"],
gtest: true,
- host_supported: true,
test_suites: ["device-tests"],
srcs: [
"EffectPreprocessingTest.cpp",
@@ -23,46 +22,18 @@
static_libs: [
"libaudiopreprocessing",
"libaudioutils",
- "webrtc_audio_processing",
],
- shared_libs: [
- "liblog",
- ],
- header_libs: [
- "libaudioeffects",
- "libhardware_headers",
- ],
- target: {
- darwin: {
- enabled: false,
- },
- },
}
cc_test {
name: "AudioPreProcessingTest",
- vendor: true,
- host_supported: true,
+ defaults: ["libaudiopreprocessing-defaults"],
gtest: false,
srcs: ["PreProcessingTest.cpp"],
- shared_libs: [
- "libaudioutils",
- "liblog",
- "libutils",
- ],
static_libs: [
"libaudiopreprocessing",
- "webrtc_audio_processing",
+ "libaudioutils",
],
- header_libs: [
- "libaudioeffects",
- "libhardware_headers",
- ],
- target: {
- darwin: {
- enabled: false,
- },
- },
}
cc_test {
diff --git a/media/libeffects/preprocessing/tests/EffectTestHelper.h b/media/libeffects/preprocessing/tests/EffectTestHelper.h
index db06823..117cf7b 100644
--- a/media/libeffects/preprocessing/tests/EffectTestHelper.h
+++ b/media/libeffects/preprocessing/tests/EffectTestHelper.h
@@ -88,7 +88,8 @@
static constexpr size_t kNumChMasks = std::size(kChMasks);
- static constexpr size_t kSampleRates[] = {8000, 16000, 24000, 32000, 48000};
+ static constexpr size_t kSampleRates[] = {8000, 11025, 12000, 16000, 22050,
+ 24000, 32000, 44100, 48000};
static constexpr size_t kNumSampleRates = std::size(kSampleRates);
diff --git a/media/libeffects/preprocessing/tests/PreProcessingTest.cpp b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
index e0025fe..3bd93f8 100644
--- a/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
+++ b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
@@ -451,8 +451,8 @@
}
audio_buffer_t inputBuffer, outputBuffer;
audio_buffer_t farInBuffer{};
- inputBuffer.frameCount = samplesRead;
- outputBuffer.frameCount = samplesRead;
+ inputBuffer.frameCount = frameLength;
+ outputBuffer.frameCount = frameLength;
inputBuffer.s16 = in.data();
outputBuffer.s16 = out.data();
@@ -472,7 +472,7 @@
}
}
- farInBuffer.frameCount = samplesRead;
+ farInBuffer.frameCount = frameLength;
farInBuffer.s16 = farIn.data();
}
@@ -519,6 +519,7 @@
}
frameCounter += frameLength;
}
+ printf("frameCounter: [%d]\n", frameCounter);
// Release all the effect handles created
for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(effectHandle[i]);
diff --git a/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh b/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh
index 942f2ec..35da13e 100755
--- a/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh
+++ b/media/libeffects/preprocessing/tests/build_and_run_all_unit_tests.sh
@@ -59,9 +59,13 @@
fs_arr=(
8000
+ 11025
+ 12000
16000
+ 22050
24000
32000
+ 44100
48000
)
diff --git a/media/libeffects/proxy/EffectProxy.cpp b/media/libeffects/proxy/EffectProxy.cpp
index c010d68..be9f8c0 100644
--- a/media/libeffects/proxy/EffectProxy.cpp
+++ b/media/libeffects/proxy/EffectProxy.cpp
@@ -116,6 +116,16 @@
pContext->sube[SUB_FX_OFFLOAD] = sube[1];
pContext->desc[SUB_FX_OFFLOAD] = desc[1];
pContext->aeli[SUB_FX_OFFLOAD] = aeli[1];
+ } else {
+ ALOGE("Both effects have (or don't have) EFFECT_FLAG_HW_ACC_TUNNEL flag");
+ delete[] sube;
+ delete[] desc;
+ delete[] aeli;
+ delete[] pContext->sube;
+ delete[] pContext->desc;
+ delete[] pContext->aeli;
+ delete pContext;
+ return -EINVAL;
}
delete[] desc;
delete[] aeli;
diff --git a/media/libmediametrics/include/MediaMetricsConstants.h b/media/libmediametrics/include/MediaMetricsConstants.h
index 84388c9..674df17 100644
--- a/media/libmediametrics/include/MediaMetricsConstants.h
+++ b/media/libmediametrics/include/MediaMetricsConstants.h
@@ -139,6 +139,7 @@
#define AMEDIAMETRICS_PROP_SESSIONID "sessionId" // int32
#define AMEDIAMETRICS_PROP_SHARINGMODE "sharingMode" // string value, "exclusive", shared"
#define AMEDIAMETRICS_PROP_SOURCE "source" // string (AudioAttributes)
+#define AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES "startThresholdFrames" // int32 (AudioTrack)
#define AMEDIAMETRICS_PROP_STARTUPMS "startupMs" // double value
// State is "ACTIVE" or "STOPPED" for AudioRecord
#define AMEDIAMETRICS_PROP_STATE "state" // string
@@ -181,6 +182,7 @@
#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETMODE "setMode" // AudioFlinger
#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE "setBufferSize" // AudioTrack
#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM "setPlaybackParam" // AudioTrack
+#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD "setStartThreshold" // AudioTrack
#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOICEVOLUME "setVoiceVolume" // AudioFlinger
#define AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME "setVolume" // AudioTrack
#define AMEDIAMETRICS_PROP_EVENT_VALUE_START "start" // AudioTrack, AudioRecord
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 7804822..472c359 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -284,8 +284,6 @@
};
sp<AudioVibrationController> mAudioVibrationController;
sp<os::ExternalVibration> mExternalVibration;
- /** How many frames should be in the buffer before the track is considered ready */
- const size_t mFrameCountToBeReady;
audio_dual_mono_mode_t mDualMonoMode = AUDIO_DUAL_MONO_MODE_OFF;
float mAudioDescriptionMixLevel = -std::numeric_limits<float>::infinity();
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index ee886d5..fb43a6e 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -638,7 +638,6 @@
mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(
uid, attr, id(), streamType, opPackageName)),
// mSinkTimestamp
- mFrameCountToBeReady(frameCountToBeReady),
mFastIndex(-1),
mCachedVolume(1.0),
/* The track might not play immediately after being active, similarly as if its volume was 0.
@@ -672,6 +671,7 @@
mFrameSize, sampleRate);
}
mServerProxy = mAudioTrackServerProxy;
+ mServerProxy->setStartThresholdInFrames(frameCountToBeReady); // update the Cblk value
// only allocate a fast track index if we were able to allocate a normal track name
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
@@ -999,7 +999,10 @@
}
size_t bufferSizeInFrames = mServerProxy->getBufferSizeInFrames();
- size_t framesToBeReady = std::min(mFrameCountToBeReady, bufferSizeInFrames);
+ // Note: mServerProxy->getStartThresholdInFrames() is clamped.
+ const size_t startThresholdInFrames = mServerProxy->getStartThresholdInFrames();
+ const size_t framesToBeReady = std::clamp( // clamp again to validate client values.
+ std::min(startThresholdInFrames, bufferSizeInFrames), size_t(1), mFrameCount);
if (framesReady() >= framesToBeReady || (mCblk->mFlags & CBLK_FORCEREADY)) {
ALOGV("%s(%d): consider track ready with %zu/%zu, target was %zu)",
@@ -1038,6 +1041,11 @@
// initial state-stopping. next state-pausing.
// What if resume is called ?
+ if (state == FLUSHED) {
+ // avoid underrun glitches when starting after flush
+ reset();
+ }
+
if (state == PAUSED || state == PAUSING) {
if (mResumeToStopping) {
// happened we need to resume to STOPPING_1