Merge "aaudio: use weak pointer to prevent UAF" into pi-dev
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/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/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/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/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/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/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/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 3736129..264e709 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -915,7 +915,7 @@
// 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->format == desc->mFormat) &&
(config->channel_mask == desc->mChannelMask) &&
(session == desc->mDirectClientSession)) {
desc->mDirectOpenCount++;
@@ -942,8 +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,
@@ -1032,7 +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)) {