Merge "Camera: fix frame capture trace pair" into pi-dev
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 936733d..61fc897 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -893,7 +893,7 @@
VideoFrame *frame = (VideoFrame *)mem->pointer();
CHECK_EQ(writeJpegFile("/sdcard/out.jpg",
- (uint8_t *)frame + sizeof(VideoFrame),
+ frame->getFlattenedData(),
frame->mWidth, frame->mHeight), 0);
}
diff --git a/drm/libmediadrm/CryptoHal.cpp b/drm/libmediadrm/CryptoHal.cpp
index 61b5127..f229751 100644
--- a/drm/libmediadrm/CryptoHal.cpp
+++ b/drm/libmediadrm/CryptoHal.cpp
@@ -22,7 +22,6 @@
#include <android/hidl/manager/1.0/IServiceManager.h>
#include <binder/IMemory.h>
-#include <cutils/native_handle.h>
#include <hidlmemory/FrameworkUtils.h>
#include <media/hardware/CryptoAPI.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -245,11 +244,6 @@
ALOGE("setHeapBase(): heap is NULL");
return -1;
}
- native_handle_t* nativeHandle = native_handle_create(1, 0);
- if (!nativeHandle) {
- ALOGE("setHeapBase(), failed to create native handle");
- return -1;
- }
Mutex::Autolock autoLock(mLock);
diff --git a/drm/libmediadrm/DrmHal.cpp b/drm/libmediadrm/DrmHal.cpp
index 4e8ad52..06e8487 100644
--- a/drm/libmediadrm/DrmHal.cpp
+++ b/drm/libmediadrm/DrmHal.cpp
@@ -1186,9 +1186,9 @@
DrmSessionManager::Instance()->useSession(sessionId);
- Status status = mPlugin->setCipherAlgorithm(toHidlVec(sessionId),
+ Return<Status> status = mPlugin->setCipherAlgorithm(toHidlVec(sessionId),
toHidlString(algorithm));
- return toStatusT(status);
+ return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
}
status_t DrmHal::setMacAlgorithm(Vector<uint8_t> const &sessionId,
@@ -1198,9 +1198,9 @@
DrmSessionManager::Instance()->useSession(sessionId);
- Status status = mPlugin->setMacAlgorithm(toHidlVec(sessionId),
+ Return<Status> status = mPlugin->setMacAlgorithm(toHidlVec(sessionId),
toHidlString(algorithm));
- return toStatusT(status);
+ return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
}
status_t DrmHal::encrypt(Vector<uint8_t> const &sessionId,
diff --git a/include/private/media/VideoFrame.h b/include/private/media/VideoFrame.h
index a9d4dd1..8b8824f 100644
--- a/include/private/media/VideoFrame.h
+++ b/include/private/media/VideoFrame.h
@@ -25,94 +25,32 @@
namespace android {
-// Represents a color converted (RGB-based) video frame
-// with bitmap pixels stored in FrameBuffer
+// Represents a color converted (RGB-based) video frame with bitmap
+// pixels stored in FrameBuffer.
+// In a VideoFrame struct stored in IMemory, frame data and ICC data
+// come after the VideoFrame structure. Their locations can be retrieved
+// by getFlattenedData() and getFlattenedIccData();
class VideoFrame
{
public:
// Construct a VideoFrame object with the specified parameters,
- // will allocate frame buffer if |allocate| is set to true, will
- // allocate buffer to hold ICC data if |iccData| and |iccSize|
- // indicate its presence.
+ // will calculate frame buffer size if |hasData| is set to true.
VideoFrame(uint32_t width, uint32_t height,
uint32_t displayWidth, uint32_t displayHeight,
- uint32_t angle, uint32_t bpp, bool allocate,
- const void *iccData, size_t iccSize):
+ uint32_t angle, uint32_t bpp, bool hasData, size_t iccSize):
mWidth(width), mHeight(height),
mDisplayWidth(displayWidth), mDisplayHeight(displayHeight),
mRotationAngle(angle), mBytesPerPixel(bpp), mRowBytes(bpp * width),
- mSize(0), mIccSize(0), mReserved(0), mData(0), mIccData(0) {
- if (allocate) {
- mSize = mRowBytes * mHeight;
- mData = new uint8_t[mSize];
- if (mData == NULL) {
- mSize = 0;
- }
- }
-
- if (iccData != NULL && iccSize > 0) {
- mIccSize = iccSize;
- mIccData = new uint8_t[iccSize];
- if (mIccData != NULL) {
- memcpy(mIccData, iccData, iccSize);
- } else {
- mIccSize = 0;
- }
- }
+ mSize(hasData ? (bpp * width * height) : 0),
+ mIccSize(iccSize), mReserved(0) {
}
- // Deep copy of both the information fields and the frame data
- VideoFrame(const VideoFrame& copy) {
- copyInfoOnly(copy);
-
- mSize = copy.mSize;
- mData = NULL; // initialize it first
- if (mSize > 0 && copy.mData != NULL) {
- mData = new uint8_t[mSize];
- if (mData != NULL) {
- memcpy(mData, copy.mData, mSize);
- } else {
- mSize = 0;
- }
- }
-
- mIccSize = copy.mIccSize;
- mIccData = NULL; // initialize it first
- if (mIccSize > 0 && copy.mIccData != NULL) {
- mIccData = new uint8_t[mIccSize];
- if (mIccData != NULL) {
- memcpy(mIccData, copy.mIccData, mIccSize);
- } else {
- mIccSize = 0;
- }
- }
- }
-
- ~VideoFrame() {
- if (mData != 0) {
- delete[] mData;
- }
- if (mIccData != 0) {
- delete[] mIccData;
- }
- }
-
- // Copy |copy| to a flattened VideoFrame in IMemory, 'this' must point to
- // a chunk of memory back by IMemory of size at least getFlattenedSize()
- // of |copy|.
- void copyFlattened(const VideoFrame& copy) {
- copyInfoOnly(copy);
-
- mSize = copy.mSize;
- mData = NULL; // initialize it first
- if (copy.mSize > 0 && copy.mData != NULL) {
- memcpy(getFlattenedData(), copy.mData, copy.mSize);
- }
-
- mIccSize = copy.mIccSize;
- mIccData = NULL; // initialize it first
- if (copy.mIccSize > 0 && copy.mIccData != NULL) {
- memcpy(getFlattenedIccData(), copy.mIccData, copy.mIccSize);
+ void init(const VideoFrame& copy, const void* iccData, size_t iccSize) {
+ *this = copy;
+ if (mIccSize == iccSize && iccSize > 0 && iccData != NULL) {
+ memcpy(getFlattenedIccData(), iccData, iccSize);
+ } else {
+ mIccSize = 0;
}
}
@@ -139,35 +77,9 @@
int32_t mRotationAngle; // Rotation angle, clockwise, should be multiple of 90
uint32_t mBytesPerPixel; // Number of bytes per pixel
uint32_t mRowBytes; // Number of bytes per row before rotation
- uint32_t mSize; // Number of bytes in mData
- uint32_t mIccSize; // Number of bytes in mIccData
+ uint32_t mSize; // Number of bytes of frame data
+ uint32_t mIccSize; // Number of bytes of ICC data
uint32_t mReserved; // (padding to make mData 64-bit aligned)
-
- // mData should be 64-bit aligned to prevent additional padding
- uint8_t* mData; // Actual binary data
- // pad structure so it's the same size on 64-bit and 32-bit
- char mPadding[8 - sizeof(mData)];
-
- // mIccData should be 64-bit aligned to prevent additional padding
- uint8_t* mIccData; // Actual binary data
- // pad structure so it's the same size on 64-bit and 32-bit
- char mIccPadding[8 - sizeof(mIccData)];
-
-private:
- //
- // Utility methods used only within VideoFrame struct
- //
-
- // Copy the information fields only
- void copyInfoOnly(const VideoFrame& copy) {
- mWidth = copy.mWidth;
- mHeight = copy.mHeight;
- mDisplayWidth = copy.mDisplayWidth;
- mDisplayHeight = copy.mDisplayHeight;
- mRotationAngle = copy.mRotationAngle;
- mBytesPerPixel = copy.mBytesPerPixel;
- mRowBytes = copy.mRowBytes;
- }
};
}; // namespace android
diff --git a/media/extractors/mkv/MatroskaExtractor.cpp b/media/extractors/mkv/MatroskaExtractor.cpp
index 1826cc1..d657582 100644
--- a/media/extractors/mkv/MatroskaExtractor.cpp
+++ b/media/extractors/mkv/MatroskaExtractor.cpp
@@ -360,7 +360,15 @@
res = mCluster->Parse(pos, len);
ALOGV("Parse (2) returned %ld", res);
- CHECK_GE(res, 0);
+
+ if (res < 0) {
+ // I/O error
+
+ ALOGE("Cluster::Parse returned result %ld", res);
+
+ mCluster = NULL;
+ break;
+ }
mBlockEntryIndex = 0;
continue;
diff --git a/media/extractors/mp4/MPEG4Extractor.cpp b/media/extractors/mp4/MPEG4Extractor.cpp
index dad36ec..ce5ca63 100644
--- a/media/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/extractors/mp4/MPEG4Extractor.cpp
@@ -628,7 +628,7 @@
track->meta.setInt32(kKeyTrackID, imageIndex);
track->includes_expensive_metadata = false;
track->skipTrack = false;
- track->timescale = 0;
+ track->timescale = 1000000;
}
}
diff --git a/media/img_utils/include/img_utils/DngUtils.h b/media/img_utils/include/img_utils/DngUtils.h
index 1d8df9c..de8f120 100644
--- a/media/img_utils/include/img_utils/DngUtils.h
+++ b/media/img_utils/include/img_utils/DngUtils.h
@@ -39,11 +39,16 @@
*/
class ANDROID_API OpcodeListBuilder : public LightRefBase<OpcodeListBuilder> {
public:
+ // Note that the Adobe DNG 1.4 spec for Bayer phase (defined for the
+ // FixBadPixelsConstant and FixBadPixelsList opcodes) is incorrect. It's
+ // inconsistent with the DNG SDK (cf. dng_negative::SetBayerMosaic and
+ // dng_opcode_FixBadPixelsList::IsGreen), and Adobe confirms that the
+ // spec should be updated to match the SDK.
enum CfaLayout {
- CFA_RGGB = 0,
- CFA_GRBG,
- CFA_GBRG,
+ CFA_GRBG = 0,
+ CFA_RGGB,
CFA_BGGR,
+ CFA_GBRG,
};
OpcodeListBuilder();
diff --git a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
index 2623697..f618d3d 100644
--- a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
+++ b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
@@ -495,7 +495,7 @@
}
void printStatus() override {
- printf("state = %d, echo gain = %f ", mState, mEchoGain);
+ printf("st = %d, echo gain = %f ", mState, mEchoGain);
}
static void sendImpulse(float *outputData, int outputChannelCount) {
@@ -670,7 +670,7 @@
printf(LOOPBACK_RESULT_TAG "phase.offset = %7.5f\n", mPhaseOffset);
printf(LOOPBACK_RESULT_TAG "ref.phase = %7.5f\n", mPhase);
printf(LOOPBACK_RESULT_TAG "frames.accumulated = %6d\n", mFramesAccumulated);
- printf(LOOPBACK_RESULT_TAG "sine.period = %6d\n", mPeriod);
+ printf(LOOPBACK_RESULT_TAG "sine.period = %6d\n", mSinePeriod);
printf(LOOPBACK_RESULT_TAG "test.state = %6d\n", mState);
printf(LOOPBACK_RESULT_TAG "frame.count = %6d\n", mFrameCounter);
// Did we ever get a lock?
@@ -684,7 +684,7 @@
}
void printStatus() override {
- printf(" state = %d, glitches = %3d,", mState, mGlitchCount);
+ printf("st = %d, #gl = %3d,", mState, mGlitchCount);
}
double calculateMagnitude(double *phasePtr = NULL) {
@@ -709,6 +709,8 @@
void process(float *inputData, int inputChannelCount,
float *outputData, int outputChannelCount,
int numFrames) override {
+ mProcessCount++;
+
float peak = measurePeakAmplitude(inputData, inputChannelCount, numFrames);
if (peak > mPeakAmplitude) {
mPeakAmplitude = peak;
@@ -720,6 +722,7 @@
float sinOut = sinf(mPhase);
switch (mState) {
+ case STATE_IDLE:
case STATE_IMMUNE:
case STATE_WAITING_FOR_SIGNAL:
break;
@@ -728,7 +731,7 @@
mCosAccumulator += sample * cosf(mPhase);
mFramesAccumulated++;
// Must be a multiple of the period or the calculation will not be accurate.
- if (mFramesAccumulated == mPeriod * 4) {
+ if (mFramesAccumulated == mSinePeriod * PERIODS_NEEDED_FOR_LOCK) {
mPhaseOffset = 0.0;
mMagnitude = calculateMagnitude(&mPhaseOffset);
if (mMagnitude > mThreshold) {
@@ -754,7 +757,22 @@
// mFrameCounter, mGlitchCount, predicted, sample);
mState = STATE_IMMUNE;
//printf("%5d: switch to STATE_IMMUNE\n", mFrameCounter);
- mDownCounter = mPeriod; // Set duration of IMMUNE state.
+ mDownCounter = mSinePeriod; // Set duration of IMMUNE state.
+ }
+
+ // Track incoming signal and slowly adjust magnitude to account
+ // for drift in the DRC or AGC.
+ mSinAccumulator += sample * sinOut;
+ mCosAccumulator += sample * cosf(mPhase);
+ mFramesAccumulated++;
+ // Must be a multiple of the period or the calculation will not be accurate.
+ if (mFramesAccumulated == mSinePeriod) {
+ const double coefficient = 0.1;
+ double phaseOffset = 0.0;
+ double magnitude = calculateMagnitude(&phaseOffset);
+ // One pole averaging filter.
+ mMagnitude = (mMagnitude * (1.0 - coefficient)) + (magnitude * coefficient);
+ resetAccumulator();
}
} break;
}
@@ -775,6 +793,9 @@
// Do these once per buffer.
switch (mState) {
+ case STATE_IDLE:
+ mState = STATE_IMMUNE; // so we can tell when
+ break;
case STATE_IMMUNE:
mDownCounter -= numFrames;
if (mDownCounter <= 0) {
@@ -805,21 +826,29 @@
void reset() override {
mGlitchCount = 0;
mState = STATE_IMMUNE;
- mPhaseIncrement = 2.0 * M_PI / mPeriod;
- printf("phaseInc = %f for period %d\n", mPhaseIncrement, mPeriod);
+ mDownCounter = IMMUNE_FRAME_COUNT;
+ mPhaseIncrement = 2.0 * M_PI / mSinePeriod;
+ printf("phaseInc = %f for period %d\n", mPhaseIncrement, mSinePeriod);
resetAccumulator();
+ mProcessCount = 0;
}
private:
enum sine_state_t {
+ STATE_IDLE,
STATE_IMMUNE,
STATE_WAITING_FOR_SIGNAL,
STATE_WAITING_FOR_LOCK,
STATE_LOCKED
};
- int mPeriod = 79;
+ enum constants {
+ IMMUNE_FRAME_COUNT = 48 * 500,
+ PERIODS_NEEDED_FOR_LOCK = 8
+ };
+
+ int mSinePeriod = 79;
double mPhaseIncrement = 0.0;
double mPhase = 0.0;
double mPhaseOffset = 0.0;
@@ -828,18 +857,19 @@
double mThreshold = 0.005;
double mTolerance = 0.01;
int32_t mFramesAccumulated = 0;
+ int32_t mProcessCount = 0;
double mSinAccumulator = 0.0;
double mCosAccumulator = 0.0;
int32_t mGlitchCount = 0;
double mPeakAmplitude = 0.0;
- int mDownCounter = 4000;
+ int mDownCounter = IMMUNE_FRAME_COUNT;
int32_t mFrameCounter = 0;
float mOutputAmplitude = 0.75;
PseudoRandom mWhiteNoise;
float mNoiseAmplitude = 0.00; // Used to experiment with warbling caused by DRC.
- sine_state_t mState = STATE_IMMUNE;
+ sine_state_t mState = STATE_IDLE;
};
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index 0ebcdbd..26d1e4b 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -38,21 +38,25 @@
// Tag for machine readable results as property = value pairs
#define RESULT_TAG "RESULT: "
#define NUM_SECONDS 5
+#define PERIOD_MILLIS 1000
#define NUM_INPUT_CHANNELS 1
#define FILENAME_ALL "/data/loopback_all.wav"
#define FILENAME_ECHOS "/data/loopback_echos.wav"
-#define APP_VERSION "0.2.01"
+#define APP_VERSION "0.2.03"
+
+constexpr int kNumCallbacksToDrain = 20;
+constexpr int kNumCallbacksToDiscard = 20;
struct LoopbackData {
AAudioStream *inputStream = nullptr;
int32_t inputFramesMaximum = 0;
int16_t *inputShortData = nullptr;
float *inputFloatData = nullptr;
- int16_t peakShort = 0;
aaudio_format_t actualInputFormat = AAUDIO_FORMAT_INVALID;
int32_t actualInputChannelCount = 0;
int32_t actualOutputChannelCount = 0;
- int32_t inputBuffersToDiscard = 10;
+ int32_t numCallbacksToDrain = kNumCallbacksToDrain;
+ int32_t numCallbacksToDiscard = kNumCallbacksToDiscard;
int32_t minNumFrames = INT32_MAX;
int32_t maxNumFrames = 0;
int32_t insufficientReadCount = 0;
@@ -131,23 +135,32 @@
myData->minNumFrames = numFrames;
}
- if (myData->inputBuffersToDiscard > 0) {
+ // Silence the output.
+ int32_t numBytes = numFrames * myData->actualOutputChannelCount * sizeof(float);
+ memset(audioData, 0 /* value */, numBytes);
+
+ if (myData->numCallbacksToDrain > 0) {
// Drain the input.
do {
framesRead = readFormattedData(myData, numFrames);
- if (framesRead < 0) {
- result = AAUDIO_CALLBACK_RESULT_STOP;
- } else if (framesRead > 0) {
- myData->inputBuffersToDiscard--;
- }
+ // Ignore errors because input stream may not be started yet.
} while (framesRead > 0);
+ myData->numCallbacksToDrain--;
- // Silence the output.
- int32_t numBytes = numFrames * myData->actualOutputChannelCount * sizeof(float);
- memset(audioData, 0 /* value */, numBytes);
+ } else if (myData->numCallbacksToDiscard > 0) {
+ // Ignore. Allow the input to fill back up to equilibrium with the output.
+ framesRead = readFormattedData(myData, numFrames);
+ if (framesRead < 0) {
+ result = AAUDIO_CALLBACK_RESULT_STOP;
+ }
+ myData->numCallbacksToDiscard--;
} else {
+ int32_t numInputBytes = numFrames * myData->actualInputChannelCount * sizeof(float);
+ memset(myData->inputFloatData, 0 /* value */, numInputBytes);
+
+ // Process data after equilibrium.
int64_t inputFramesWritten = AAudioStream_getFramesWritten(myData->inputStream);
int64_t inputFramesRead = AAudioStream_getFramesRead(myData->inputStream);
int64_t framesAvailable = inputFramesWritten - inputFramesRead;
@@ -155,6 +168,7 @@
if (framesRead < 0) {
result = AAUDIO_CALLBACK_RESULT_STOP;
} else {
+
if (framesRead < numFrames) {
if(framesRead < (int32_t) framesAvailable) {
printf("insufficient but numFrames = %d, framesRead = %d, available = %d\n",
@@ -172,13 +186,13 @@
// Save for later.
myData->audioRecording.write(myData->inputFloatData,
myData->actualInputChannelCount,
- framesRead);
+ numFrames);
// Analyze the data.
myData->loopbackProcessor->process(myData->inputFloatData,
myData->actualInputChannelCount,
outputData,
myData->actualOutputChannelCount,
- framesRead);
+ numFrames);
myData->isDone = myData->loopbackProcessor->isDone();
if (myData->isDone) {
result = AAUDIO_CALLBACK_RESULT_STOP;
@@ -366,7 +380,9 @@
}
int32_t requestedDuration = argParser.getDurationSeconds();
- int32_t recordingDuration = std::min(60, requestedDuration);
+ int32_t requestedDurationMillis = requestedDuration * MILLIS_PER_SECOND;
+ int32_t timeMillis = 0;
+ int32_t recordingDuration = std::min(60 * 5, requestedDuration);
switch(testMode) {
case TEST_SINE_MAGNITUDE:
@@ -449,7 +465,6 @@
// Allocate a buffer for the audio data.
loopbackData.inputFramesMaximum = 32 * AAudioStream_getFramesPerBurst(inputStream);
- loopbackData.inputBuffersToDiscard = 200;
if (loopbackData.actualInputFormat == AAUDIO_FORMAT_PCM_I16) {
loopbackData.inputShortData = new int16_t[loopbackData.inputFramesMaximum
@@ -460,13 +475,7 @@
loopbackData.loopbackProcessor->reset();
- result = recorder.start();
- if (result != AAUDIO_OK) {
- printf("ERROR - AAudioStream_requestStart(input) returned %d = %s\n",
- result, AAudio_convertResultToText(result));
- goto finish;
- }
-
+ // Start OUTPUT first so INPUT does not overflow.
result = player.start();
if (result != AAUDIO_OK) {
printf("ERROR - AAudioStream_requestStart(output) returned %d = %s\n",
@@ -474,9 +483,15 @@
goto finish;
}
- printf("------- sleep while the callback runs --------------\n");
- fflush(stdout);
- for (int i = requestedDuration; i > 0 ; i--) {
+ result = recorder.start();
+ if (result != AAUDIO_OK) {
+ printf("ERROR - AAudioStream_requestStart(input) returned %d = %s\n",
+ result, AAudio_convertResultToText(result));
+ goto finish;
+ }
+
+ printf("------- sleep and log while the callback runs --------------\n");
+ while (timeMillis <= requestedDurationMillis) {
if (loopbackData.inputError != AAUDIO_OK) {
printf(" ERROR on input stream\n");
break;
@@ -487,10 +502,9 @@
printf(" test says it is done!\n");
break;
} else {
- sleep(1);
- printf("%4d: ", i);
+ // Log a line of stream data.
+ printf("%7.3f: ", 0.001 * timeMillis); // display in seconds
loopbackData.loopbackProcessor->printStatus();
-
printf(" insf %3d,", (int) loopbackData.insufficientReadCount);
int64_t inputFramesWritten = AAudioStream_getFramesWritten(inputStream);
@@ -498,7 +512,7 @@
int64_t outputFramesWritten = AAudioStream_getFramesWritten(outputStream);
int64_t outputFramesRead = AAudioStream_getFramesRead(outputStream);
static const int textOffset = strlen("AAUDIO_STREAM_STATE_"); // strip this off
- printf(" INPUT: wr %7lld - rd %7lld = %5lld, state %s, oruns %3d | ",
+ printf(" | INPUT: wr %7lld - rd %7lld = %5lld, st %8s, oruns %3d",
(long long) inputFramesWritten,
(long long) inputFramesRead,
(long long) (inputFramesWritten - inputFramesRead),
@@ -506,7 +520,7 @@
AAudioStream_getState(inputStream))[textOffset],
AAudioStream_getXRunCount(inputStream));
- printf(" OUTPUT: wr %7lld - rd %7lld = %5lld, state %s, uruns %3d\n",
+ printf(" | OUTPUT: wr %7lld - rd %7lld = %5lld, st %8s, uruns %3d\n",
(long long) outputFramesWritten,
(long long) outputFramesRead,
(long long) (outputFramesWritten - outputFramesRead),
@@ -515,6 +529,9 @@
AAudioStream_getXRunCount(outputStream)
);
}
+ int32_t periodMillis = (timeMillis < 2000) ? PERIOD_MILLIS / 4 : PERIOD_MILLIS;
+ usleep(periodMillis * 1000);
+ timeMillis += periodMillis;
}
result = player.stop();
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index 4a2a0a8..40ebb76 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -59,10 +59,15 @@
return (int16_t) roundf(fmaxf(fminf(f * scale, scale - 1.f), -scale));
}
+// Clip to valid range of a float sample to prevent excessive volume.
+// By using fmin and fmax we also protect against NaN.
+static float clipToMinMaxHeadroom(float input) {
+ return fmin(MAX_HEADROOM, fmax(MIN_HEADROOM, input));
+}
+
static float clipAndClampFloatToPcm16(float sample, float scaler) {
// Clip to valid range of a float sample to prevent excessive volume.
- if (sample > MAX_HEADROOM) sample = MAX_HEADROOM;
- else if (sample < MIN_HEADROOM) sample = MIN_HEADROOM;
+ sample = clipToMinMaxHeadroom(sample);
// Scale and convert to a short.
float fval = sample * scaler;
@@ -127,6 +132,7 @@
}
}
+
// This code assumes amplitude1 and amplitude2 are between 0.0 and 1.0
void AAudio_linearRamp(const float *source,
float *destination,
@@ -139,10 +145,8 @@
for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
float sample = *source++;
-
// Clip to valid range of a float sample to prevent excessive volume.
- if (sample > MAX_HEADROOM) sample = MAX_HEADROOM;
- else if (sample < MIN_HEADROOM) sample = MIN_HEADROOM;
+ sample = clipToMinMaxHeadroom(sample);
*destination++ = sample * scaler;
}
@@ -240,8 +244,7 @@
float sample = *source++;
// Clip to valid range of a float sample to prevent excessive volume.
- if (sample > MAX_HEADROOM) sample = MAX_HEADROOM;
- else if (sample < MIN_HEADROOM) sample = MIN_HEADROOM;
+ sample = clipToMinMaxHeadroom(sample);
const float scaler = amplitude1 + (frameIndex * delta);
float sampleScaled = sample * scaler;
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 1c4a80e..39ee437 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -20,6 +20,7 @@
#include <utils/Log.h>
#include <binder/IServiceManager.h>
#include <binder/ProcessState.h>
+#include <binder/IPCThreadState.h>
#include <media/AudioResamplerPublic.h>
#include <media/AudioSystem.h>
#include <media/IAudioFlinger.h>
@@ -75,7 +76,9 @@
af = gAudioFlinger;
}
if (afc != 0) {
+ int64_t token = IPCThreadState::self()->clearCallingIdentity();
af->registerClient(afc);
+ IPCThreadState::self()->restoreCallingIdentity(token);
}
return af;
}
@@ -767,7 +770,10 @@
ap = gAudioPolicyService;
}
if (apc != 0) {
+ int64_t token = IPCThreadState::self()->clearCallingIdentity();
ap->registerClient(apc);
+ ap->setAudioPortCallbacksEnabled(apc->isAudioPortCbEnabled());
+ IPCThreadState::self()->restoreCallingIdentity(token);
}
return ap;
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 8c9d3c1..00af7e8 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -871,7 +871,6 @@
switch (code) {
case SET_STREAM_VOLUME:
case SET_STREAM_MUTE:
- case SET_MODE:
case OPEN_OUTPUT:
case OPEN_DUPLICATE_OUTPUT:
case CLOSE_OUTPUT:
@@ -892,7 +891,15 @@
case SET_RECORD_SILENCED:
ALOGW("%s: transaction %d received from PID %d",
__func__, code, IPCThreadState::self()->getCallingPid());
- return INVALID_OPERATION;
+ // return status only for non void methods
+ switch (code) {
+ case SET_RECORD_SILENCED:
+ break;
+ default:
+ reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
+ break;
+ }
+ return OK;
default:
break;
}
@@ -909,7 +916,15 @@
ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
__func__, code, IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid());
- return INVALID_OPERATION;
+ // return status only for non void methods
+ switch (code) {
+ case SYSTEM_READY:
+ break;
+ default:
+ reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
+ break;
+ }
+ return OK;
}
} break;
default:
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index 35f9727..abb502b 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -857,7 +857,16 @@
case RELEASE_SOUNDTRIGGER_SESSION:
ALOGW("%s: transaction %d received from PID %d",
__func__, code, IPCThreadState::self()->getCallingPid());
- return INVALID_OPERATION;
+ // return status only for non void methods
+ switch (code) {
+ case RELEASE_OUTPUT:
+ case RELEASE_INPUT:
+ break;
+ default:
+ reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
+ break;
+ }
+ return OK;
default:
break;
}
@@ -867,7 +876,6 @@
case SET_DEVICE_CONNECTION_STATE:
case HANDLE_DEVICE_CONFIG_CHANGE:
case SET_PHONE_STATE:
- case SET_RINGER_MODE:
case SET_FORCE_USE:
case INIT_STREAM_VOLUME:
case SET_STREAM_VOLUME:
@@ -879,7 +887,8 @@
ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
__func__, code, IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid());
- return INVALID_OPERATION;
+ reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
+ return OK;
}
} break;
default:
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 22b700d..0c4d6ee 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -431,6 +431,7 @@
int addAudioPortCallback(const sp<AudioPortCallback>& callback);
int removeAudioPortCallback(const sp<AudioPortCallback>& callback);
+ bool isAudioPortCbEnabled() const { return (mAudioPortCallbacks.size() != 0); }
// DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
diff --git a/media/libmedia/NdkWrapper.cpp b/media/libmedia/NdkWrapper.cpp
index 6f56d0c..272bc30 100644
--- a/media/libmedia/NdkWrapper.cpp
+++ b/media/libmedia/NdkWrapper.cpp
@@ -31,6 +31,18 @@
#include <media/stagefright/foundation/AMessage.h>
#include <utils/Errors.h>
+// TODO: remove forward declaration when AMediaExtractor_disconnect is offcially added to NDK
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+media_status_t AMediaExtractor_disconnect(AMediaExtractor *);
+
+#ifdef __cplusplus
+} // extern "C"
+#endif
+
namespace android {
static const size_t kAESBlockSize = 16; // AES_BLOCK_SIZE
diff --git a/media/libmedia/include/media/CodecServiceRegistrant.h b/media/libmedia/include/media/CodecServiceRegistrant.h
new file mode 100644
index 0000000..e0af781
--- /dev/null
+++ b/media/libmedia/include/media/CodecServiceRegistrant.h
@@ -0,0 +1,23 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CODEC_SERVICE_REGISTRANT_H_
+
+#define CODEC_SERVICE_REGISTRANT_H_
+
+typedef void (*RegisterCodecServicesFunc)();
+
+#endif // CODEC_SERVICE_REGISTRANT_H_
diff --git a/media/libmedia/include/media/MediaMetadataRetrieverInterface.h b/media/libmedia/include/media/MediaMetadataRetrieverInterface.h
index c45a964..992e230 100644
--- a/media/libmedia/include/media/MediaMetadataRetrieverInterface.h
+++ b/media/libmedia/include/media/MediaMetadataRetrieverInterface.h
@@ -43,12 +43,12 @@
virtual status_t setDataSource(int fd, int64_t offset, int64_t length) = 0;
virtual status_t setDataSource(const sp<DataSource>& source, const char *mime) = 0;
- virtual VideoFrame* getFrameAtTime(
+ virtual sp<IMemory> getFrameAtTime(
int64_t timeUs, int option, int colorFormat, bool metaOnly) = 0;
- virtual VideoFrame* getImageAtIndex(
+ virtual sp<IMemory> getImageAtIndex(
int index, int colorFormat, bool metaOnly, bool thumbnail) = 0;
virtual status_t getFrameAtIndex(
- std::vector<VideoFrame*>* frames,
+ std::vector<sp<IMemory> >* frames,
int frameIndex, int numFrames, int colorFormat, bool metaOnly) = 0;
virtual MediaAlbumArt* extractAlbumArt() = 0;
virtual const char* extractMetadata(int keyCode) = 0;
@@ -61,14 +61,14 @@
MediaMetadataRetrieverInterface() {}
virtual ~MediaMetadataRetrieverInterface() {}
- virtual VideoFrame* getFrameAtTime(
+ virtual sp<IMemory> getFrameAtTime(
int64_t /*timeUs*/, int /*option*/, int /*colorFormat*/, bool /*metaOnly*/)
{ return NULL; }
- virtual VideoFrame* getImageAtIndex(
+ virtual sp<IMemory> getImageAtIndex(
int /*index*/, int /*colorFormat*/, bool /*metaOnly*/, bool /*thumbnail*/)
{ return NULL; }
virtual status_t getFrameAtIndex(
- std::vector<VideoFrame*>* /*frames*/,
+ std::vector<sp<IMemory> >* /*frames*/,
int /*frameIndex*/, int /*numFrames*/, int /*colorFormat*/, bool /*metaOnly*/)
{ return ERROR_UNSUPPORTED; }
virtual MediaAlbumArt* extractAlbumArt() { return NULL; }
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.cpp b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
index 3b3ac29..672b832 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.cpp
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
@@ -194,25 +194,6 @@
Mutex MetadataRetrieverClient::sLock;
-static sp<IMemory> getThumbnail(VideoFrame* frame) {
- std::unique_ptr<VideoFrame> frameDeleter(frame);
-
- size_t size = frame->getFlattenedSize();
- sp<MemoryHeapBase> heap = new MemoryHeapBase(size, 0, "MetadataRetrieverClient");
- if (heap == NULL) {
- ALOGE("failed to create MemoryDealer");
- return NULL;
- }
- sp<IMemory> thrumbnail = new MemoryBase(heap, 0, size);
- if (thrumbnail == NULL) {
- ALOGE("not enough memory for VideoFrame size=%zu", size);
- return NULL;
- }
- VideoFrame *frameCopy = static_cast<VideoFrame *>(thrumbnail->pointer());
- frameCopy->copyFlattened(*frame);
- return thrumbnail;
-}
-
sp<IMemory> MetadataRetrieverClient::getFrameAtTime(
int64_t timeUs, int option, int colorFormat, bool metaOnly)
{
@@ -225,12 +206,12 @@
ALOGE("retriever is not initialized");
return NULL;
}
- VideoFrame *frame = mRetriever->getFrameAtTime(timeUs, option, colorFormat, metaOnly);
+ sp<IMemory> frame = mRetriever->getFrameAtTime(timeUs, option, colorFormat, metaOnly);
if (frame == NULL) {
ALOGE("failed to capture a video frame");
return NULL;
}
- return getThumbnail(frame);
+ return frame;
}
sp<IMemory> MetadataRetrieverClient::getImageAtIndex(
@@ -244,12 +225,12 @@
ALOGE("retriever is not initialized");
return NULL;
}
- VideoFrame *frame = mRetriever->getImageAtIndex(index, colorFormat, metaOnly, thumbnail);
+ sp<IMemory> frame = mRetriever->getImageAtIndex(index, colorFormat, metaOnly, thumbnail);
if (frame == NULL) {
ALOGE("failed to extract image");
return NULL;
}
- return getThumbnail(frame);
+ return frame;
}
status_t MetadataRetrieverClient::getFrameAtIndex(
@@ -264,15 +245,12 @@
return INVALID_OPERATION;
}
- std::vector<VideoFrame*> videoFrames;
status_t err = mRetriever->getFrameAtIndex(
- &videoFrames, frameIndex, numFrames, colorFormat, metaOnly);
+ frames, frameIndex, numFrames, colorFormat, metaOnly);
if (err != OK) {
+ frames->clear();
return err;
}
- for (size_t i = 0; i < videoFrames.size(); i++) {
- frames->push_back(getThumbnail(videoFrames[i]));
- }
return OK;
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index a762e76..3a28bbd 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -1703,6 +1703,8 @@
++mAudioDrainGeneration;
if (mAudioRenderingStartGeneration != -1) {
prepareForMediaRenderingStart_l();
+ // PauseTimeout is applied to offload mode only. Cancel pending timer.
+ cancelAudioOffloadPauseTimeout();
}
}
@@ -1805,6 +1807,12 @@
if (mAudioTornDown) {
return;
}
+
+ // TimeoutWhenPaused is only for offload mode.
+ if (reason == kDueToTimeout && !offloadingAudio()) {
+ return;
+ }
+
mAudioTornDown = true;
int64_t currentPositionUs;
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index ad81f04..2abea9e 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -2144,6 +2144,10 @@
// value is unknown
drc.targetRefLevel = -1;
}
+ if (!msg->findInt32("aac-drc-effect-type", &drc.effectType)) {
+ // value is unknown
+ drc.effectType = -2; // valid values are -1 and over
+ }
err = setupAACCodec(
encoder, numChannels, sampleRate, bitrate, aacProfile,
@@ -2778,7 +2782,7 @@
? OMX_AUDIO_AACStreamFormatMP4ADTS
: OMX_AUDIO_AACStreamFormatMP4FF;
- OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE presentation;
+ OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE presentation;
InitOMXParams(&presentation);
presentation.nMaxOutputChannels = maxOutputChannelCount;
presentation.nDrcCut = drc.drcCut;
@@ -2787,14 +2791,29 @@
presentation.nTargetReferenceLevel = drc.targetRefLevel;
presentation.nEncodedTargetLevel = drc.encodedTargetLevel;
presentation.nPCMLimiterEnable = pcmLimiterEnable;
+ presentation.nDrcEffectType = drc.effectType;
status_t res = mOMXNode->setParameter(
OMX_IndexParamAudioAac, &profile, sizeof(profile));
if (res == OK) {
// optional parameters, will not cause configuration failure
- mOMXNode->setParameter(
+ if (mOMXNode->setParameter(
+ (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAacDrcPresentation,
+ &presentation, sizeof(presentation)) == ERROR_UNSUPPORTED) {
+ // prior to 9.0 we used a different config structure and index
+ OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE presentation8;
+ InitOMXParams(&presentation8);
+ presentation8.nMaxOutputChannels = presentation.nMaxOutputChannels;
+ presentation8.nDrcCut = presentation.nDrcCut;
+ presentation8.nDrcBoost = presentation.nDrcBoost;
+ presentation8.nHeavyCompression = presentation.nHeavyCompression;
+ presentation8.nTargetReferenceLevel = presentation.nTargetReferenceLevel;
+ presentation8.nEncodedTargetLevel = presentation.nEncodedTargetLevel;
+ presentation8.nPCMLimiterEnable = presentation.nPCMLimiterEnable;
+ (void)mOMXNode->setParameter(
(OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAacPresentation,
- &presentation, sizeof(presentation));
+ &presentation8, sizeof(presentation8));
+ }
} else {
ALOGW("did not set AudioAndroidAacPresentation due to error %d when setting AudioAac", res);
}
diff --git a/media/libstagefright/FrameDecoder.cpp b/media/libstagefright/FrameDecoder.cpp
index a00d13a..09a8be5 100644
--- a/media/libstagefright/FrameDecoder.cpp
+++ b/media/libstagefright/FrameDecoder.cpp
@@ -17,12 +17,11 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "FrameDecoder"
-#include <inttypes.h>
-
-#include <utils/Log.h>
-#include <gui/Surface.h>
-
#include "include/FrameDecoder.h"
+#include <binder/MemoryBase.h>
+#include <binder/MemoryHeapBase.h>
+#include <gui/Surface.h>
+#include <inttypes.h>
#include <media/ICrypto.h>
#include <media/IMediaSource.h>
#include <media/MediaCodecBuffer.h>
@@ -36,14 +35,15 @@
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/Utils.h>
#include <private/media/VideoFrame.h>
+#include <utils/Log.h>
namespace android {
-static const int64_t kBufferTimeOutUs = 30000ll; // 30 msec
-static const size_t kRetryCount = 20; // must be >0
+static const int64_t kBufferTimeOutUs = 10000ll; // 10 msec
+static const size_t kRetryCount = 50; // must be >0
//static
-VideoFrame *allocVideoFrame(const sp<MetaData> &trackMeta,
+sp<IMemory> allocVideoFrame(const sp<MetaData>& trackMeta,
int32_t width, int32_t height, int32_t dstBpp, bool metaOnly = false) {
int32_t rotationAngle;
if (!trackMeta->findInt32(kKeyRotation, &rotationAngle)) {
@@ -74,8 +74,24 @@
displayHeight = height;
}
- return new VideoFrame(width, height, displayWidth, displayHeight,
- rotationAngle, dstBpp, !metaOnly, iccData, iccSize);
+ VideoFrame frame(width, height, displayWidth, displayHeight,
+ rotationAngle, dstBpp, !metaOnly, iccSize);
+
+ size_t size = frame.getFlattenedSize();
+ sp<MemoryHeapBase> heap = new MemoryHeapBase(size, 0, "MetadataRetrieverClient");
+ if (heap == NULL) {
+ ALOGE("failed to create MemoryDealer");
+ return NULL;
+ }
+ sp<IMemory> frameMem = new MemoryBase(heap, 0, size);
+ if (frameMem == NULL) {
+ ALOGE("not enough memory for VideoFrame size=%zu", size);
+ return NULL;
+ }
+ VideoFrame* frameCopy = static_cast<VideoFrame*>(frameMem->pointer());
+ frameCopy->init(frame, iccData, iccSize);
+
+ return frameMem;
}
//static
@@ -92,7 +108,7 @@
}
//static
-VideoFrame* FrameDecoder::getMetadataOnly(
+sp<IMemory> FrameDecoder::getMetadataOnly(
const sp<MetaData> &trackMeta, int colorFormat, bool thumbnail) {
OMX_COLOR_FORMATTYPE dstFormat;
int32_t dstBpp;
@@ -146,7 +162,7 @@
return false;
}
-VideoFrame* FrameDecoder::extractFrame(
+sp<IMemory> FrameDecoder::extractFrame(
int64_t frameTimeUs, int option, int colorFormat) {
if (!getDstColorFormat(
(android_pixel_format_t)colorFormat, &mDstFormat, &mDstBpp)) {
@@ -158,12 +174,12 @@
return NULL;
}
- return mFrames.size() > 0 ? mFrames[0].release() : NULL;
+ return mFrames.size() > 0 ? mFrames[0] : NULL;
}
status_t FrameDecoder::extractFrames(
int64_t frameTimeUs, size_t numFrames, int option, int colorFormat,
- std::vector<VideoFrame*>* frames) {
+ std::vector<sp<IMemory> >* frames) {
if (!getDstColorFormat(
(android_pixel_format_t)colorFormat, &mDstFormat, &mDstBpp)) {
return ERROR_UNSUPPORTED;
@@ -175,7 +191,7 @@
}
for (size_t i = 0; i < mFrames.size(); i++) {
- frames->push_back(mFrames[i].release());
+ frames->push_back(mFrames[i]);
}
return OK;
}
@@ -253,10 +269,13 @@
uint32_t flags = 0;
sp<MediaCodecBuffer> codecBuffer = NULL;
+ // Queue as many inputs as we possibly can, then block on dequeuing
+ // outputs. After getting each output, come back and queue the inputs
+ // again to keep the decoder busy.
while (haveMoreInputs) {
- err = decoder->dequeueInputBuffer(&inputIndex, kBufferTimeOutUs);
+ err = decoder->dequeueInputBuffer(&inputIndex, 0);
if (err != OK) {
- ALOGW("Timed out waiting for input");
+ ALOGV("Timed out waiting for input");
if (retriesLeft) {
err = OK;
}
@@ -295,27 +314,21 @@
}
mediaBuffer->release();
- break;
- }
- if (haveMoreInputs && inputIndex < inputBuffers.size()) {
- ALOGV("QueueInput: size=%zu ts=%" PRId64 " us flags=%x",
- codecBuffer->size(), ptsUs, flags);
+ if (haveMoreInputs && inputIndex < inputBuffers.size()) {
+ ALOGV("QueueInput: size=%zu ts=%" PRId64 " us flags=%x",
+ codecBuffer->size(), ptsUs, flags);
- err = decoder->queueInputBuffer(
- inputIndex,
- codecBuffer->offset(),
- codecBuffer->size(),
- ptsUs,
- flags);
+ err = decoder->queueInputBuffer(
+ inputIndex,
+ codecBuffer->offset(),
+ codecBuffer->size(),
+ ptsUs,
+ flags);
- if (flags & MediaCodec::BUFFER_FLAG_EOS) {
- haveMoreInputs = false;
- }
-
- // we don't expect an output from codec config buffer
- if (flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) {
- continue;
+ if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+ haveMoreInputs = false;
+ }
}
}
@@ -468,12 +481,13 @@
crop_bottom = height - 1;
}
- VideoFrame *frame = allocVideoFrame(
+ sp<IMemory> frameMem = allocVideoFrame(
trackMeta(),
(crop_right - crop_left + 1),
(crop_bottom - crop_top + 1),
dstBpp());
- addFrame(frame);
+ addFrame(frameMem);
+ VideoFrame* frame = static_cast<VideoFrame*>(frameMem->pointer());
int32_t srcFormat;
CHECK(outputFormat->findInt32("color-format", &srcFormat));
@@ -485,7 +499,7 @@
(const uint8_t *)videoFrameBuffer->data(),
width, height,
crop_left, crop_top, crop_right, crop_bottom,
- frame->mData,
+ frame->getFlattenedData(),
frame->mWidth,
frame->mHeight,
crop_left, crop_top, crop_right, crop_bottom);
@@ -596,9 +610,10 @@
}
if (mFrame == NULL) {
- mFrame = allocVideoFrame(trackMeta(), imageWidth, imageHeight, dstBpp());
+ sp<IMemory> frameMem = allocVideoFrame(trackMeta(), imageWidth, imageHeight, dstBpp());
+ mFrame = static_cast<VideoFrame*>(frameMem->pointer());
- addFrame(mFrame);
+ addFrame(frameMem);
}
int32_t srcFormat;
@@ -639,7 +654,7 @@
(const uint8_t *)videoFrameBuffer->data(),
width, height,
crop_left, crop_top, crop_right, crop_bottom,
- mFrame->mData,
+ mFrame->getFlattenedData(),
mFrame->mWidth,
mFrame->mHeight,
dstLeft, dstTop, dstRight, dstBottom);
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index a3261d7..550a99c 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -496,7 +496,6 @@
mStreamableFile = false;
mTimeScale = -1;
mHasFileLevelMeta = false;
- mHasMoovBox = false;
mPrimaryItemId = 0;
mAssociationEntryCount = 0;
mNumGrids = 0;
@@ -505,6 +504,7 @@
// And they will stay the same for all the recording sessions.
if (isFirstSession) {
mMoovExtraSize = 0;
+ mHasMoovBox = false;
mMetaKeys = new AMessage();
addDeviceMeta();
mLatitudex10000 = 0;
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 23bee49..06a49d0 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -764,7 +764,7 @@
// ignore stuff with no presentation time
if (presentationUs <= 0) {
- ALOGD("-- returned buffer has bad timestamp %" PRId64 ", ignore it", presentationUs);
+ ALOGV("-- returned buffer timestamp %" PRId64 " <= 0, ignore it", presentationUs);
mLatencyUnknown++;
return;
}
diff --git a/media/libstagefright/StagefrightMediaScanner.cpp b/media/libstagefright/StagefrightMediaScanner.cpp
index 4ff2bfe..e010b3e 100644
--- a/media/libstagefright/StagefrightMediaScanner.cpp
+++ b/media/libstagefright/StagefrightMediaScanner.cpp
@@ -40,7 +40,8 @@
".mpeg", ".ogg", ".mid", ".smf", ".imy", ".wma", ".aac",
".wav", ".amr", ".midi", ".xmf", ".rtttl", ".rtx", ".ota",
".mkv", ".mka", ".webm", ".ts", ".fl", ".flac", ".mxmf",
- ".avi", ".mpeg", ".mpg", ".awb", ".mpga", ".mov"
+ ".avi", ".mpeg", ".mpg", ".awb", ".mpga", ".mov",
+ ".m4v", ".oga"
};
static const size_t kNumValidExtensions =
sizeof(kValidExtensions) / sizeof(kValidExtensions[0]);
@@ -62,6 +63,11 @@
client.setLocale(locale());
client.beginFile();
MediaScanResult result = processFileInternal(path, mimeType, client);
+ ALOGV("result: %d", result);
+ if (mimeType == NULL && result != MEDIA_SCAN_RESULT_OK) {
+ ALOGW("media scan failed for %s", path);
+ client.setMimeType("application/octet-stream");
+ }
client.endFile();
return result;
}
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index 6ad6004..5417fef 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -124,7 +124,7 @@
return OK;
}
-VideoFrame* StagefrightMetadataRetriever::getImageAtIndex(
+sp<IMemory> StagefrightMetadataRetriever::getImageAtIndex(
int index, int colorFormat, bool metaOnly, bool thumbnail) {
ALOGV("getImageAtIndex: index(%d) colorFormat(%d) metaOnly(%d) thumbnail(%d)",
@@ -193,7 +193,7 @@
for (size_t i = 0; i < matchingCodecs.size(); ++i) {
const AString &componentName = matchingCodecs[i];
ImageDecoder decoder(componentName, trackMeta, source);
- VideoFrame* frame = decoder.extractFrame(
+ sp<IMemory> frame = decoder.extractFrame(
thumbnail ? -1 : 0 /*frameTimeUs*/, 0 /*seekMode*/, colorFormat);
if (frame != NULL) {
@@ -205,19 +205,19 @@
return NULL;
}
-VideoFrame* StagefrightMetadataRetriever::getFrameAtTime(
+sp<IMemory> StagefrightMetadataRetriever::getFrameAtTime(
int64_t timeUs, int option, int colorFormat, bool metaOnly) {
ALOGV("getFrameAtTime: %" PRId64 " us option: %d colorFormat: %d, metaOnly: %d",
timeUs, option, colorFormat, metaOnly);
- VideoFrame *frame;
+ sp<IMemory> frame;
status_t err = getFrameInternal(
timeUs, 1, option, colorFormat, metaOnly, &frame, NULL /*outFrames*/);
return (err == OK) ? frame : NULL;
}
status_t StagefrightMetadataRetriever::getFrameAtIndex(
- std::vector<VideoFrame*>* frames,
+ std::vector<sp<IMemory> >* frames,
int frameIndex, int numFrames, int colorFormat, bool metaOnly) {
ALOGV("getFrameAtIndex: frameIndex %d, numFrames %d, colorFormat: %d, metaOnly: %d",
frameIndex, numFrames, colorFormat, metaOnly);
@@ -229,7 +229,7 @@
status_t StagefrightMetadataRetriever::getFrameInternal(
int64_t timeUs, int numFrames, int option, int colorFormat, bool metaOnly,
- VideoFrame **outFrame, std::vector<VideoFrame*>* outFrames) {
+ sp<IMemory>* outFrame, std::vector<sp<IMemory> >* outFrames) {
if (mExtractor.get() == NULL) {
ALOGE("no extractor.");
return NO_INIT;
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index c61f4b5..cf5e91e 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -154,6 +154,7 @@
{ 23, OMX_AUDIO_AACObjectLD },
{ 29, OMX_AUDIO_AACObjectHE_PS },
{ 39, OMX_AUDIO_AACObjectELD },
+ { 42, OMX_AUDIO_AACObjectXHE },
};
OMX_AUDIO_AACPROFILETYPE profile;
@@ -1610,6 +1611,7 @@
{ OMX_AUDIO_AACObjectLD, AUDIO_FORMAT_AAC_LD},
{ OMX_AUDIO_AACObjectHE_PS, AUDIO_FORMAT_AAC_HE_V2},
{ OMX_AUDIO_AACObjectELD, AUDIO_FORMAT_AAC_ELD},
+ { OMX_AUDIO_AACObjectXHE, AUDIO_FORMAT_AAC_XHE},
{ OMX_AUDIO_AACObjectNull, AUDIO_FORMAT_AAC},
};
diff --git a/media/libstagefright/codecs/aacdec/DrcPresModeWrap.cpp b/media/libstagefright/codecs/aacdec/DrcPresModeWrap.cpp
index 129ad65..95d3724 100644
--- a/media/libstagefright/codecs/aacdec/DrcPresModeWrap.cpp
+++ b/media/libstagefright/codecs/aacdec/DrcPresModeWrap.cpp
@@ -24,7 +24,7 @@
//#define DRC_PRES_MODE_WRAP_DEBUG
#define GPM_ENCODER_TARGET_LEVEL 64
-#define MAX_TARGET_LEVEL 64
+#define MAX_TARGET_LEVEL 40
CDrcPresModeWrapper::CDrcPresModeWrapper()
{
@@ -164,7 +164,7 @@
if (mDataUpdate) {
// sanity check
if (mDesTarget < MAX_TARGET_LEVEL){
- mDesTarget = MAX_TARGET_LEVEL; // limit target level to -16 dB or below
+ mDesTarget = MAX_TARGET_LEVEL; // limit target level to -10 dB or below
newTarget = MAX_TARGET_LEVEL;
}
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index e0c0c32..ecd2512 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -36,6 +36,7 @@
#define DRC_DEFAULT_MOBILE_DRC_CUT 127 /* maximum compression of dynamic range for mobile conf */
#define DRC_DEFAULT_MOBILE_DRC_BOOST 127 /* maximum compression of dynamic range for mobile conf */
#define DRC_DEFAULT_MOBILE_DRC_HEAVY 1 /* switch for heavy compression for mobile conf */
+#define DRC_DEFAULT_MOBILE_DRC_EFFECT 3 /* MPEG-D DRC effect type; 3 => Limited playback range */
#define DRC_DEFAULT_MOBILE_ENC_LEVEL (-1) /* encoder target level; -1 => the value is unknown, otherwise dB step value (e.g. 64 for -16 dB) */
#define MAX_CHANNEL_COUNT 8 /* maximum number of audio channels that can be decoded */
// names of properties that can be used to override the default DRC settings
@@ -44,6 +45,7 @@
#define PROP_DRC_OVERRIDE_BOOST "aac_drc_boost"
#define PROP_DRC_OVERRIDE_HEAVY "aac_drc_heavy"
#define PROP_DRC_OVERRIDE_ENC_LEVEL "aac_drc_enc_target_level"
+#define PROP_DRC_OVERRIDE_EFFECT "aac_drc_effect_type"
namespace android {
@@ -207,6 +209,17 @@
} else {
mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET, DRC_DEFAULT_MOBILE_ENC_LEVEL);
}
+ // AAC_UNIDRC_SET_EFFECT
+ int32_t effectType = DRC_DEFAULT_MOBILE_DRC_EFFECT;
+ // FIXME can't read default property for DRC effect type
+ //int32_t effectType =
+ // property_get_int32(PROP_DRC_OVERRIDE_EFFECT, DRC_DEFAULT_MOBILE_DRC_EFFECT);
+ if (effectType < -1 || effectType > 8) {
+ effectType = DRC_DEFAULT_MOBILE_DRC_EFFECT;
+ }
+ ALOGV("AAC decoder using MPEG-D DRC effect type %d (default=%d)",
+ effectType, DRC_DEFAULT_MOBILE_DRC_EFFECT);
+ aacDecoder_SetParam(mAACDecoder, AAC_UNIDRC_SET_EFFECT, effectType);
// By default, the decoder creates a 5.1 channel downmix signal.
// For seven and eight channel input streams, enable 6.1 and 7.1 channel output
@@ -414,10 +427,10 @@
return OMX_ErrorNone;
}
- case OMX_IndexParamAudioAndroidAacPresentation:
+ case OMX_IndexParamAudioAndroidAacDrcPresentation:
{
- const OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE *aacPresParams =
- (const OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE *)params;
+ const OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE *aacPresParams =
+ (const OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE *)params;
if (!isValidOMXParam(aacPresParams)) {
return OMX_ErrorBadParameter;
@@ -443,6 +456,10 @@
ALOGV("set nMaxOutputChannels=%d", max);
aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, max);
}
+ if (aacPresParams->nDrcEffectType >= -1) {
+ ALOGV("set nDrcEffectType=%d", aacPresParams->nDrcEffectType);
+ aacDecoder_SetParam(mAACDecoder, AAC_UNIDRC_SET_EFFECT, aacPresParams->nDrcEffectType);
+ }
bool updateDrcWrapper = false;
if (aacPresParams->nDrcBoost >= 0) {
ALOGV("set nDrcBoost=%d", aacPresParams->nDrcBoost);
diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 8f349fc..9bdf895 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -484,6 +484,9 @@
// Base URL must be absolute
return false;
}
+ if (!strncasecmp("data:", url, 5)) {
+ return false;
+ }
const size_t schemeEnd = (strstr(baseURL, "//") - baseURL) + 2;
CHECK(schemeEnd == 7 || schemeEnd == 8);
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 5624f4a..f292c47 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -33,6 +33,7 @@
#include <media/stagefright/foundation/ByteUtils.h>
#include <media/stagefright/foundation/MediaKeys.h>
#include <media/stagefright/foundation/avc_utils.h>
+#include <media/stagefright/DataURISource.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MetaDataUtils.h>
@@ -347,6 +348,16 @@
sp<ABuffer> key;
if (index >= 0) {
key = mAESKeyForURI.valueAt(index);
+ } else if (keyURI.startsWith("data:")) {
+ sp<DataSource> keySrc = DataURISource::Create(keyURI.c_str());
+ off64_t keyLen;
+ if (keySrc == NULL || keySrc->getSize(&keyLen) != OK || keyLen < 0) {
+ ALOGE("Malformed cipher key data uri.");
+ return ERROR_MALFORMED;
+ }
+ key = new ABuffer(keyLen);
+ keySrc->readAt(0, key->data(), keyLen);
+ key->setRange(0, keyLen);
} else {
ssize_t err = mHTTPDownloader->fetchFile(keyURI.c_str(), &key);
diff --git a/media/libstagefright/include/FrameDecoder.h b/media/libstagefright/include/FrameDecoder.h
index b67e928..f6d4727 100644
--- a/media/libstagefright/include/FrameDecoder.h
+++ b/media/libstagefright/include/FrameDecoder.h
@@ -44,16 +44,16 @@
mDstFormat(OMX_COLOR_Format16bitRGB565),
mDstBpp(2) {}
- VideoFrame* extractFrame(int64_t frameTimeUs, int option, int colorFormat);
+ sp<IMemory> extractFrame(int64_t frameTimeUs, int option, int colorFormat);
status_t extractFrames(
int64_t frameTimeUs,
size_t numFrames,
int option,
int colorFormat,
- std::vector<VideoFrame*>* frames);
+ std::vector<sp<IMemory> >* frames);
- static VideoFrame* getMetadataOnly(
+ static sp<IMemory> getMetadataOnly(
const sp<MetaData> &trackMeta, int colorFormat, bool thumbnail = false);
protected:
@@ -81,8 +81,8 @@
OMX_COLOR_FORMATTYPE dstFormat() const { return mDstFormat; }
int32_t dstBpp() const { return mDstBpp; }
- void addFrame(VideoFrame *frame) {
- mFrames.push_back(std::unique_ptr<VideoFrame>(frame));
+ void addFrame(const sp<IMemory> &frame) {
+ mFrames.push_back(frame);
}
private:
@@ -91,7 +91,7 @@
sp<IMediaSource> mSource;
OMX_COLOR_FORMATTYPE mDstFormat;
int32_t mDstBpp;
- std::vector<std::unique_ptr<VideoFrame> > mFrames;
+ std::vector<sp<IMemory> > mFrames;
static bool getDstColorFormat(
android_pixel_format_t colorFormat,
diff --git a/media/libstagefright/include/StagefrightMetadataRetriever.h b/media/libstagefright/include/StagefrightMetadataRetriever.h
index 8443fbe..209f850 100644
--- a/media/libstagefright/include/StagefrightMetadataRetriever.h
+++ b/media/libstagefright/include/StagefrightMetadataRetriever.h
@@ -40,12 +40,12 @@
virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
virtual status_t setDataSource(const sp<DataSource>& source, const char *mime);
- virtual VideoFrame* getFrameAtTime(
+ virtual sp<IMemory> getFrameAtTime(
int64_t timeUs, int option, int colorFormat, bool metaOnly);
- virtual VideoFrame* getImageAtIndex(
+ virtual sp<IMemory> getImageAtIndex(
int index, int colorFormat, bool metaOnly, bool thumbnail);
virtual status_t getFrameAtIndex(
- std::vector<VideoFrame*>* frames,
+ std::vector<sp<IMemory> >* frames,
int frameIndex, int numFrames, int colorFormat, bool metaOnly);
virtual MediaAlbumArt *extractAlbumArt();
@@ -65,7 +65,7 @@
status_t getFrameInternal(
int64_t timeUs, int numFrames, int option, int colorFormat, bool metaOnly,
- VideoFrame **outFrame, std::vector<VideoFrame*>* outFrames);
+ sp<IMemory>* outFrame, std::vector<sp<IMemory> >* outFrames);
StagefrightMetadataRetriever(const StagefrightMetadataRetriever &);
diff --git a/media/libstagefright/include/media/stagefright/ACodec.h b/media/libstagefright/include/media/stagefright/ACodec.h
index 64caeed..97d15a7 100644
--- a/media/libstagefright/include/media/stagefright/ACodec.h
+++ b/media/libstagefright/include/media/stagefright/ACodec.h
@@ -446,6 +446,7 @@
int32_t heavyCompression;
int32_t targetRefLevel;
int32_t encodedTargetLevel;
+ int32_t effectType;
} drcParams_t;
status_t setupAACCodec(
diff --git a/media/ndk/NdkMediaDrm.cpp b/media/ndk/NdkMediaDrm.cpp
index fe08ab9..6d10f1c 100644
--- a/media/ndk/NdkMediaDrm.cpp
+++ b/media/ndk/NdkMediaDrm.cpp
@@ -224,7 +224,7 @@
static bool findId(AMediaDrm *mObj, const AMediaDrmByteArray &id, List<idvec_t>::iterator &iter) {
for (iter = mObj->mIds.begin(); iter != mObj->mIds.end(); ++iter) {
- if (iter->array() == id.ptr && iter->size() == id.length) {
+ if (id.length == iter->size() && memcmp(iter->array(), id.ptr, iter->size()) == 0) {
return true;
}
}
diff --git a/media/ndk/include/media/NdkMediaExtractor.h b/media/ndk/include/media/NdkMediaExtractor.h
index 1d295e4..f7b9cfd 100644
--- a/media/ndk/include/media/NdkMediaExtractor.h
+++ b/media/ndk/include/media/NdkMediaExtractor.h
@@ -216,12 +216,6 @@
#endif /* __ANDROID_API__ >= 28 */
-#if __ANDROID_API__ >= 29
-
-media_status_t AMediaExtractor_disconnect(AMediaExtractor *ex);
-
-#endif /* __ANDROID_API__ >= 29 */
-
#endif /* __ANDROID_API__ >= 21 */
__END_DECLS
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index ef466a2..79bb9fe 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -216,11 +216,6 @@
mWarmupNsMax = LONG_MAX;
}
mMixerBufferState = UNDEFINED;
-#if !LOG_NDEBUG
- for (unsigned i = 0; i < FastMixerState::sMaxFastTracks; ++i) {
- mFastTrackNames[i] = -1;
- }
-#endif
// we need to reconfigure all active tracks
previousTrackMask = 0;
mFastTracksGen = current->mFastTracksGen - 1;
@@ -245,9 +240,6 @@
if (mMixer != NULL) {
mMixer->destroy(i);
}
-#if !LOG_NDEBUG
- mFastTrackNames[i] = -1;
-#endif
// don't reset track dump state, since other side is ignoring it
mGenerations[i] = fastTrack->mGeneration;
}
@@ -259,7 +251,6 @@
addedTracks &= ~(1 << i);
const FastTrack* fastTrack = ¤t->mFastTracks[i];
AudioBufferProvider *bufferProvider = fastTrack->mBufferProvider;
- ALOG_ASSERT(bufferProvider != NULL && mFastTrackNames[i] == -1);
if (mMixer != NULL) {
const int name = i; // for clarity, choose name as fast track index.
status_t status = mMixer->create(
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 20de97c..c47aa01 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -6603,8 +6603,17 @@
if (mPipeSource != 0) {
size_t framesToRead = mBufferSize / mFrameSize;
framesToRead = min(mRsmpInFramesOA - rear, mRsmpInFramesP2 / 2);
- framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize,
- framesToRead);
+
+ // The audio fifo read() returns OVERRUN on overflow, and advances the read pointer
+ // to the full buffer point (clearing the overflow condition). Upon OVERRUN error,
+ // we immediately retry the read() to get data and prevent another overflow.
+ for (int retries = 0; retries <= 2; ++retries) {
+ ALOGW_IF(retries > 0, "overrun on read from pipe, retry #%d", retries);
+ framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize,
+ framesToRead);
+ if (framesRead != OVERRUN) break;
+ }
+
// since pipe is non-blocking, simulate blocking input by waiting for 1/2 of
// buffer size or at least for 20ms.
size_t sleepFrames = max(
diff --git a/services/audiopolicy/common/managerdefinitions/include/SessionRoute.h b/services/audiopolicy/common/managerdefinitions/include/SessionRoute.h
index fc2c273..fac6cbe 100644
--- a/services/audiopolicy/common/managerdefinitions/include/SessionRoute.h
+++ b/services/audiopolicy/common/managerdefinitions/include/SessionRoute.h
@@ -54,7 +54,7 @@
void log(const char* prefix);
- bool isActive() {
+ bool isActiveOrChanged() {
return (mDeviceDescriptor != 0) && (mChanged || (mActivityCount > 0));
}
@@ -96,7 +96,7 @@
int incRouteActivity(audio_session_t session);
int decRouteActivity(audio_session_t session);
- bool hasRouteChanged(audio_session_t session); // also clears the changed flag
+ bool getAndClearRouteChanged(audio_session_t session); // also clears the changed flag
void log(const char* caption);
// Specify an Output(Sink) route by passing SessionRoute::SOURCE_TYPE_NA in the
diff --git a/services/audiopolicy/common/managerdefinitions/src/SessionRoute.cpp b/services/audiopolicy/common/managerdefinitions/src/SessionRoute.cpp
index 689f4e6..8edd4d1 100644
--- a/services/audiopolicy/common/managerdefinitions/src/SessionRoute.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/SessionRoute.cpp
@@ -40,7 +40,7 @@
return indexOfKey(session) >= 0 && valueFor(session)->mDeviceDescriptor != 0;
}
-bool SessionRouteMap::hasRouteChanged(audio_session_t session)
+bool SessionRouteMap::getAndClearRouteChanged(audio_session_t session)
{
if (indexOfKey(session) >= 0) {
if (valueFor(session)->mChanged) {
@@ -104,9 +104,7 @@
sp<SessionRoute> route = indexOfKey(session) >= 0 ? valueFor(session) : 0;
if (route != 0) {
- if (((route->mDeviceDescriptor == 0) && (descriptor != 0)) ||
- ((route->mDeviceDescriptor != 0) &&
- ((descriptor == 0) || (!route->mDeviceDescriptor->equals(descriptor))))) {
+ if (descriptor != 0 || route->mDeviceDescriptor != 0) {
route->mChanged = true;
}
route->mRefCount++;
@@ -114,10 +112,10 @@
} else {
route = new SessionRoute(session, streamType, source, descriptor, uid);
route->mRefCount++;
- add(session, route);
if (descriptor != 0) {
route->mChanged = true;
}
+ add(session, route);
}
}
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index bb00c3f..264e709 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -822,7 +822,7 @@
"flags %#x",
device, config->sample_rate, config->format, config->channel_mask, *flags);
- *output = getOutputForDevice(device, session, *stream, *output, config, flags);
+ *output = getOutputForDevice(device, session, *stream, config, flags);
if (*output == AUDIO_IO_HANDLE_NONE) {
mOutputRoutes.removeRoute(session);
return INVALID_OPERATION;
@@ -841,11 +841,10 @@
audio_devices_t device,
audio_session_t session,
audio_stream_type_t stream,
- audio_io_handle_t originalOutput,
const audio_config_t *config,
audio_output_flags_t *flags)
{
- audio_io_handle_t output = originalOutput;
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status;
// open a direct output if required by specified parameters
@@ -909,22 +908,20 @@
}
if (profile != 0) {
- // exclude MMAP streams
- if ((*flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0 || output != AUDIO_IO_HANDLE_NONE) {
- for (size_t i = 0; i < mOutputs.size(); i++) {
- sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
- if (!desc->isDuplicated() && (profile == desc->mProfile)) {
- // reuse direct output if currently open by the same client
- // and configured with same parameters
- if ((config->sample_rate == desc->mSamplingRate) &&
- audio_formats_match(config->format, desc->mFormat) &&
- (config->channel_mask == desc->mChannelMask) &&
- (session == desc->mDirectClientSession)) {
- desc->mDirectOpenCount++;
- ALOGI("getOutputForDevice() reusing direct output %d for session %d",
- mOutputs.keyAt(i), session);
- return mOutputs.keyAt(i);
- }
+ // exclusive outputs for MMAP and Offload are enforced by different session ids.
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+ // reuse direct output if currently open by the same client
+ // and configured with same parameters
+ if ((config->sample_rate == desc->mSamplingRate) &&
+ (config->format == desc->mFormat) &&
+ (config->channel_mask == desc->mChannelMask) &&
+ (session == desc->mDirectClientSession)) {
+ desc->mDirectOpenCount++;
+ ALOGI("getOutputForDevice() reusing direct output %d for session %d",
+ mOutputs.keyAt(i), session);
+ return mOutputs.keyAt(i);
}
}
}
@@ -945,8 +942,7 @@
// only accept an output with the requested parameters
if (status != NO_ERROR ||
(config->sample_rate != 0 && config->sample_rate != outputDesc->mSamplingRate) ||
- (config->format != AUDIO_FORMAT_DEFAULT &&
- !audio_formats_match(config->format, outputDesc->mFormat)) ||
+ (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->mFormat) ||
(config->channel_mask != 0 && config->channel_mask != outputDesc->mChannelMask)) {
ALOGV("getOutputForDevice() failed opening direct output: output %d sample rate %d %d,"
"format %d %d, channel mask %04x %04x", output, config->sample_rate,
@@ -1035,7 +1031,7 @@
// if a valid format is specified, skip output if not compatible
if (format != AUDIO_FORMAT_INVALID) {
if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
- if (!audio_formats_match(format, outputDesc->mFormat)) {
+ if (format != outputDesc->mFormat) {
continue;
}
} else if (!audio_is_linear_pcm(format)) {
@@ -1116,7 +1112,7 @@
} else {
newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
}
- } else if (mOutputRoutes.hasRouteChanged(session)) {
+ } else if (mOutputRoutes.getAndClearRouteChanged(session)) {
newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
checkStrategyRoute(getStrategy(stream), output);
} else {
@@ -1980,7 +1976,7 @@
// Routing?
mInputRoutes.incRouteActivity(session);
- if (audioSession->activeCount() == 1 || mInputRoutes.hasRouteChanged(session)) {
+ if (audioSession->activeCount() == 1 || mInputRoutes.getAndClearRouteChanged(session)) {
// indicate active capture to sound trigger service if starting capture from a mic on
// primary HW module
audio_devices_t device = getNewInputDevice(inputDesc);
@@ -4707,7 +4703,7 @@
for (size_t routeIndex = 0; routeIndex < mOutputRoutes.size(); routeIndex++) {
sp<SessionRoute> route = mOutputRoutes.valueAt(routeIndex);
routing_strategy routeStrategy = getStrategy(route->mStreamType);
- if ((routeStrategy == strategy) && route->isActive() &&
+ if ((routeStrategy == strategy) && route->isActiveOrChanged() &&
(mAvailableOutputDevices.indexOf(route->mDeviceDescriptor) >= 0)) {
return route->mDeviceDescriptor->type();
}
@@ -5139,7 +5135,7 @@
// then select this device.
for (size_t routeIndex = 0; routeIndex < mInputRoutes.size(); routeIndex++) {
sp<SessionRoute> route = mInputRoutes.valueAt(routeIndex);
- if ((inputSource == route->mSource) && route->isActive() &&
+ if ((inputSource == route->mSource) && route->isActiveOrChanged() &&
(mAvailableInputDevices.indexOf(route->mDeviceDescriptor) >= 0)) {
return route->mDeviceDescriptor->type();
}
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index d05ba1f..2b68882 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -628,7 +628,6 @@
audio_devices_t device,
audio_session_t session,
audio_stream_type_t stream,
- audio_io_handle_t originalOutput,
const audio_config_t *config,
audio_output_flags_t *flags);
// internal method to return the input handle for the given device and format
diff --git a/services/camera/libcameraservice/common/CameraProviderManager.cpp b/services/camera/libcameraservice/common/CameraProviderManager.cpp
index 077e05e..66e9196 100644
--- a/services/camera/libcameraservice/common/CameraProviderManager.cpp
+++ b/services/camera/libcameraservice/common/CameraProviderManager.cpp
@@ -783,6 +783,18 @@
name.c_str(), statusToString(status));
return nullptr;
}
+
+ for (auto& conflictName : resourceCost.conflictingDevices) {
+ uint16_t major, minor;
+ std::string type, id;
+ status_t res = parseDeviceName(conflictName, &major, &minor, &type, &id);
+ if (res != OK) {
+ ALOGE("%s: Failed to parse conflicting device %s", __FUNCTION__, conflictName.c_str());
+ return nullptr;
+ }
+ conflictName = id;
+ }
+
return std::unique_ptr<DeviceInfo>(
new DeviceInfoT(name, tagId, id, minorVersion, resourceCost,
cameraInterface));
diff --git a/services/mediacodec/main_codecservice.cpp b/services/mediacodec/main_codecservice.cpp
index 701ca6e..51619f6 100644
--- a/services/mediacodec/main_codecservice.cpp
+++ b/services/mediacodec/main_codecservice.cpp
@@ -25,6 +25,9 @@
#include <media/stagefright/omx/1.0/Omx.h>
#include <media/stagefright/omx/1.0/OmxStore.h>
+#include <media/CodecServiceRegistrant.h>
+#include <dlfcn.h>
+
using namespace android;
// Must match location in Android.mk.
@@ -45,20 +48,37 @@
::android::hardware::configureRpcThreadpool(64, false);
- using namespace ::android::hardware::media::omx::V1_0;
- sp<IOmxStore> omxStore = new implementation::OmxStore();
- if (omxStore == nullptr) {
- LOG(ERROR) << "Cannot create IOmxStore HAL service.";
- } else if (omxStore->registerAsService() != OK) {
- LOG(ERROR) << "Cannot register IOmxStore HAL service.";
- }
- sp<IOmx> omx = new implementation::Omx();
- if (omx == nullptr) {
- LOG(ERROR) << "Cannot create IOmx HAL service.";
- } else if (omx->registerAsService() != OK) {
- LOG(ERROR) << "Cannot register IOmx HAL service.";
+ // Registration of customized codec services
+ void *registrantLib = dlopen(
+ "libmedia_codecserviceregistrant.so",
+ RTLD_NOW | RTLD_LOCAL);
+ if (registrantLib) {
+ RegisterCodecServicesFunc registerCodecServices =
+ reinterpret_cast<RegisterCodecServicesFunc>(
+ dlsym(registrantLib, "RegisterCodecServices"));
+ if (registerCodecServices) {
+ registerCodecServices();
+ } else {
+ LOG(WARNING) << "Cannot register additional services "
+ "-- corrupted library.";
+ }
} else {
- LOG(INFO) << "IOmx HAL service created.";
+ // Default codec services
+ using namespace ::android::hardware::media::omx::V1_0;
+ sp<IOmxStore> omxStore = new implementation::OmxStore();
+ if (omxStore == nullptr) {
+ LOG(ERROR) << "Cannot create IOmxStore HAL service.";
+ } else if (omxStore->registerAsService() != OK) {
+ LOG(ERROR) << "Cannot register IOmxStore HAL service.";
+ }
+ sp<IOmx> omx = new implementation::Omx();
+ if (omx == nullptr) {
+ LOG(ERROR) << "Cannot create IOmx HAL service.";
+ } else if (omx->registerAsService() != OK) {
+ LOG(ERROR) << "Cannot register IOmx HAL service.";
+ } else {
+ LOG(INFO) << "IOmx HAL service created.";
+ }
}
::android::hardware::joinRpcThreadpool();
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
index 6015b28..253f290 100644
--- a/services/oboeservice/AAudioServiceEndpoint.h
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -49,9 +49,9 @@
virtual aaudio_result_t close() = 0;
- virtual aaudio_result_t registerStream(android::sp<AAudioServiceStreamBase> stream);
+ aaudio_result_t registerStream(android::sp<AAudioServiceStreamBase> stream);
- virtual aaudio_result_t unregisterStream(android::sp<AAudioServiceStreamBase> stream);
+ aaudio_result_t unregisterStream(android::sp<AAudioServiceStreamBase> stream);
virtual aaudio_result_t startStream(android::sp<AAudioServiceStreamBase> stream,
audio_port_handle_t *clientHandle) = 0;
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index c943008..48d8002 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -105,6 +105,9 @@
goto error;
}
+ // This is not protected by a lock because the stream cannot be
+ // referenced until the service returns a handle to the client.
+ // So only one thread can open a stream.
mServiceEndpoint = mEndpointManager.openEndpoint(mAudioService,
request,
sharingMode);
@@ -113,6 +116,9 @@
result = AAUDIO_ERROR_UNAVAILABLE;
goto error;
}
+ // Save a weak pointer that we will use to access the endpoint.
+ mServiceEndpointWeak = mServiceEndpoint;
+
mFramesPerBurst = mServiceEndpoint->getFramesPerBurst();
copyFrom(*mServiceEndpoint);
}
@@ -131,13 +137,16 @@
stop();
- if (mServiceEndpoint == nullptr) {
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
result = AAUDIO_ERROR_INVALID_STATE;
} else {
- mServiceEndpoint->unregisterStream(this);
- AAudioEndpointManager &mEndpointManager = AAudioEndpointManager::getInstance();
- mEndpointManager.closeEndpoint(mServiceEndpoint);
- mServiceEndpoint.clear();
+ endpoint->unregisterStream(this);
+ AAudioEndpointManager &endpointManager = AAudioEndpointManager::getInstance();
+ endpointManager.closeEndpoint(endpoint);
+
+ // AAudioService::closeStream() prevents two threads from closing at the same time.
+ mServiceEndpoint.clear(); // endpoint will hold the pointer until this method returns.
}
{
@@ -153,7 +162,12 @@
aaudio_result_t AAudioServiceStreamBase::startDevice() {
mClientHandle = AUDIO_PORT_HANDLE_NONE;
- return mServiceEndpoint->startStream(this, &mClientHandle);
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ return endpoint->startStream(this, &mClientHandle);
}
/**
@@ -163,16 +177,11 @@
*/
aaudio_result_t AAudioServiceStreamBase::start() {
aaudio_result_t result = AAUDIO_OK;
+
if (isRunning()) {
return AAUDIO_OK;
}
- if (mServiceEndpoint == nullptr) {
- ALOGE("%s() missing endpoint", __func__);
- result = AAUDIO_ERROR_INVALID_STATE;
- goto error;
- }
-
setFlowing(false);
// Start with fresh presentation timestamps.
@@ -201,10 +210,6 @@
if (!isRunning()) {
return result;
}
- if (mServiceEndpoint == nullptr) {
- ALOGE("%s() missing endpoint", __func__);
- return AAUDIO_ERROR_INVALID_STATE;
- }
// Send it now because the timestamp gets rounded up when stopStream() is called below.
// Also we don't need the timestamps while we are shutting down.
@@ -216,7 +221,12 @@
return result;
}
- result = mServiceEndpoint->stopStream(this, mClientHandle);
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ result = endpoint->stopStream(this, mClientHandle);
if (result != AAUDIO_OK) {
ALOGE("%s() mServiceEndpoint returned %d, %s", __func__, result, getTypeText());
disconnect(); // TODO should we return or pause Base first?
@@ -233,11 +243,6 @@
return result;
}
- if (mServiceEndpoint == nullptr) {
- ALOGE("%s() missing endpoint", __func__);
- return AAUDIO_ERROR_INVALID_STATE;
- }
-
setState(AAUDIO_STREAM_STATE_STOPPING);
// Send it now because the timestamp gets rounded up when stopStream() is called below.
@@ -249,10 +254,15 @@
return result;
}
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
// TODO wait for data to be played out
- result = mServiceEndpoint->stopStream(this, mClientHandle);
+ result = endpoint->stopStream(this, mClientHandle);
if (result != AAUDIO_OK) {
- ALOGE("%s() mServiceEndpoint returned %d, %s", __func__, result, getTypeText());
+ ALOGE("%s() stopStream returned %d, %s", __func__, result, getTypeText());
disconnect();
// TODO what to do with result here?
}
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index d8102be..0ad015e 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -279,7 +279,12 @@
SimpleDoubleBuffer<Timestamp> mAtomicTimestamp;
android::AAudioService &mAudioService;
+
+ // The mServiceEndpoint variable can be accessed by multiple threads.
+ // So we access it by locally promoting a weak pointer to a smart pointer,
+ // which is thread-safe.
android::sp<AAudioServiceEndpoint> mServiceEndpoint;
+ android::wp<AAudioServiceEndpoint> mServiceEndpointWeak;
private:
aaudio_handle_t mHandle = -1;
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.cpp b/services/oboeservice/AAudioServiceStreamMMAP.cpp
index 34ddb4b..c845309 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.cpp
+++ b/services/oboeservice/AAudioServiceStreamMMAP.cpp
@@ -70,14 +70,19 @@
return result;
}
- result = mServiceEndpoint->registerStream(keep);
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+
+ result = endpoint->registerStream(keep);
if (result != AAUDIO_OK) {
- goto error;
+ return result;
}
setState(AAUDIO_STREAM_STATE_OPEN);
-error:
return AAUDIO_OK;
}
@@ -118,21 +123,37 @@
aaudio_result_t AAudioServiceStreamMMAP::startClient(const android::AudioClient& client,
audio_port_handle_t *clientHandle) {
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
// Start the client on behalf of the application. Generate a new porthandle.
- aaudio_result_t result = mServiceEndpoint->startClient(client, clientHandle);
+ aaudio_result_t result = endpoint->startClient(client, clientHandle);
return result;
}
aaudio_result_t AAudioServiceStreamMMAP::stopClient(audio_port_handle_t clientHandle) {
- aaudio_result_t result = mServiceEndpoint->stopClient(clientHandle);
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ aaudio_result_t result = endpoint->stopClient(clientHandle);
return result;
}
// Get free-running DSP or DMA hardware position from the HAL.
aaudio_result_t AAudioServiceStreamMMAP::getFreeRunningPosition(int64_t *positionFrames,
int64_t *timeNanos) {
- sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP{
- static_cast<AAudioServiceEndpointMMAP *>(mServiceEndpoint.get())};
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP =
+ static_cast<AAudioServiceEndpointMMAP *>(endpoint.get());
+
aaudio_result_t result = serviceEndpointMMAP->getFreeRunningPosition(positionFrames, timeNanos);
if (result == AAUDIO_OK) {
Timestamp timestamp(*positionFrames, *timeNanos);
@@ -148,8 +169,15 @@
// Get timestamp that was written by getFreeRunningPosition()
aaudio_result_t AAudioServiceStreamMMAP::getHardwareTimestamp(int64_t *positionFrames,
int64_t *timeNanos) {
- sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP{
- static_cast<AAudioServiceEndpointMMAP *>(mServiceEndpoint.get())};
+
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP =
+ static_cast<AAudioServiceEndpointMMAP *>(endpoint.get());
+
// TODO Get presentation timestamp from the HAL
if (mAtomicTimestamp.isValid()) {
Timestamp timestamp = mAtomicTimestamp.read();
@@ -165,7 +193,12 @@
aaudio_result_t AAudioServiceStreamMMAP::getAudioDataDescription(
AudioEndpointParcelable &parcelable)
{
- sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP{
- static_cast<AAudioServiceEndpointMMAP *>(mServiceEndpoint.get())};
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ sp<AAudioServiceEndpointMMAP> serviceEndpointMMAP =
+ static_cast<AAudioServiceEndpointMMAP *>(endpoint.get());
return serviceEndpointMMAP->getDownDataDescription(parcelable);
}
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index 864a008..05c5735 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -128,6 +128,12 @@
const AAudioStreamConfiguration &configurationInput = request.getConstantConfiguration();
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ result = AAUDIO_ERROR_INVALID_STATE;
+ goto error;
+ }
+
// Is the request compatible with the shared endpoint?
setFormat(configurationInput.getFormat());
if (getFormat() == AAUDIO_FORMAT_UNSPECIFIED) {
@@ -140,20 +146,20 @@
setSampleRate(configurationInput.getSampleRate());
if (getSampleRate() == AAUDIO_UNSPECIFIED) {
- setSampleRate(mServiceEndpoint->getSampleRate());
- } else if (getSampleRate() != mServiceEndpoint->getSampleRate()) {
+ setSampleRate(endpoint->getSampleRate());
+ } else if (getSampleRate() != endpoint->getSampleRate()) {
ALOGD("%s() mSampleRate = %d, need %d",
- __func__, getSampleRate(), mServiceEndpoint->getSampleRate());
+ __func__, getSampleRate(), endpoint->getSampleRate());
result = AAUDIO_ERROR_INVALID_RATE;
goto error;
}
setSamplesPerFrame(configurationInput.getSamplesPerFrame());
if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) {
- setSamplesPerFrame(mServiceEndpoint->getSamplesPerFrame());
- } else if (getSamplesPerFrame() != mServiceEndpoint->getSamplesPerFrame()) {
+ setSamplesPerFrame(endpoint->getSamplesPerFrame());
+ } else if (getSamplesPerFrame() != endpoint->getSamplesPerFrame()) {
ALOGD("%s() mSamplesPerFrame = %d, need %d",
- __func__, getSamplesPerFrame(), mServiceEndpoint->getSamplesPerFrame());
+ __func__, getSamplesPerFrame(), endpoint->getSamplesPerFrame());
result = AAUDIO_ERROR_OUT_OF_RANGE;
goto error;
}
@@ -179,7 +185,10 @@
}
}
- result = mServiceEndpoint->registerStream(keep);
+ ALOGD("AAudioServiceStreamShared::open() actual rate = %d, channels = %d, deviceId = %d",
+ getSampleRate(), getSamplesPerFrame(), endpoint->getDeviceId());
+
+ result = endpoint->registerStream(keep);
if (result != AAUDIO_OK) {
goto error;
}
@@ -246,7 +255,13 @@
int64_t *timeNanos) {
int64_t position = 0;
- aaudio_result_t result = mServiceEndpoint->getTimestamp(&position, timeNanos);
+ sp<AAudioServiceEndpoint> endpoint = mServiceEndpointWeak.promote();
+ if (endpoint == nullptr) {
+ ALOGE("%s() has no endpoint", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+
+ aaudio_result_t result = endpoint->getTimestamp(&position, timeNanos);
if (result == AAUDIO_OK) {
int64_t offset = mTimestampPositionOffset.load();
// TODO, do not go below starting value