Merge "Add missing extensions, fix scan of bad media files." into pi-dev
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/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/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/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/libstagefright/FrameDecoder.cpp b/media/libstagefright/FrameDecoder.cpp
index b17fbc9..09a8be5 100644
--- a/media/libstagefright/FrameDecoder.cpp
+++ b/media/libstagefright/FrameDecoder.cpp
@@ -39,8 +39,8 @@
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
sp<IMemory> allocVideoFrame(const sp<MetaData>& trackMeta,
@@ -269,10 +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;
}
@@ -311,27 +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;
+ }
}
}
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/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 22bc426..3736129 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) &&
+ 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);
}
}
}
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