Merge "Fix green or corrupted video frames in the exported movies" into jb-dev
diff --git a/include/media/AudioParameter.h b/include/media/AudioParameter.h
index 8cb2fa7..891bc4b 100644
--- a/include/media/AudioParameter.h
+++ b/include/media/AudioParameter.h
@@ -40,12 +40,14 @@
// keyFrameCount: to change audio output frame count, value is an int
// keyInputSource: to change audio input source, value is an int in audio_source_t
// (defined in media/mediarecorder.h)
+ // keyScreenState: either "on" or "off"
static const char * const keyRouting;
static const char * const keySamplingRate;
static const char * const keyFormat;
static const char * const keyChannels;
static const char * const keyFrameCount;
static const char * const keyInputSource;
+ static const char * const keyScreenState;
String8 toString();
diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp
index 44d05cd..d3c69f4 100644
--- a/media/libeffects/visualizer/EffectVisualizer.cpp
+++ b/media/libeffects/visualizer/EffectVisualizer.cpp
@@ -52,6 +52,8 @@
// that the framework has stopped playing audio and we must start returning silence
#define MAX_STALL_TIME_MS 1000
+#define CAPTURE_BUF_SIZE 65536 // "64k should be enough for everyone"
+
struct VisualizerContext {
const struct effect_interface_s *mItfe;
effect_config_t mConfig;
@@ -59,10 +61,10 @@
uint32_t mCaptureSize;
uint32_t mScalingMode;
uint8_t mState;
- uint8_t mCurrentBuf;
- uint8_t mLastBuf;
+ uint8_t mLastCaptureIdx;
+ uint32_t mLatency;
struct timespec mBufferUpdateTime;
- uint8_t mCaptureBuf[2][VISUALIZER_CAPTURE_SIZE_MAX];
+ uint8_t mCaptureBuf[CAPTURE_BUF_SIZE];
};
//
@@ -72,11 +74,10 @@
void Visualizer_reset(VisualizerContext *pContext)
{
pContext->mCaptureIdx = 0;
- pContext->mCurrentBuf = 0;
- pContext->mLastBuf = 1;
+ pContext->mLastCaptureIdx = 0;
pContext->mBufferUpdateTime.tv_sec = 0;
- memset(pContext->mCaptureBuf[0], 0x80, VISUALIZER_CAPTURE_SIZE_MAX);
- memset(pContext->mCaptureBuf[1], 0x80, VISUALIZER_CAPTURE_SIZE_MAX);
+ pContext->mLatency = 0;
+ memset(pContext->mCaptureBuf, 0x80, CAPTURE_BUF_SIZE);
}
//----------------------------------------------------------------------------
@@ -316,25 +317,25 @@
uint32_t captIdx;
uint32_t inIdx;
- uint8_t *buf = pContext->mCaptureBuf[pContext->mCurrentBuf];
+ uint8_t *buf = pContext->mCaptureBuf;
for (inIdx = 0, captIdx = pContext->mCaptureIdx;
- inIdx < inBuffer->frameCount && captIdx < pContext->mCaptureSize;
+ inIdx < inBuffer->frameCount;
inIdx++, captIdx++) {
+ if (captIdx >= CAPTURE_BUF_SIZE) {
+ // wrap around
+ captIdx = 0;
+ }
int32_t smp = inBuffer->s16[2 * inIdx] + inBuffer->s16[2 * inIdx + 1];
smp = smp >> shift;
buf[captIdx] = ((uint8_t)smp)^0x80;
}
+
+ // XXX the following two should really be atomic, though it probably doesn't
+ // matter much for visualization purposes
pContext->mCaptureIdx = captIdx;
-
- // go to next buffer when buffer full
- if (pContext->mCaptureIdx == pContext->mCaptureSize) {
- pContext->mCurrentBuf ^= 1;
- pContext->mCaptureIdx = 0;
-
- // update last buffer update time stamp
- if (clock_gettime(CLOCK_MONOTONIC, &pContext->mBufferUpdateTime) < 0) {
- pContext->mBufferUpdateTime.tv_sec = 0;
- }
+ // update last buffer update time stamp
+ if (clock_gettime(CLOCK_MONOTONIC, &pContext->mBufferUpdateTime) < 0) {
+ pContext->mBufferUpdateTime.tv_sec = 0;
}
if (inBuffer->raw != outBuffer->raw) {
@@ -464,6 +465,10 @@
pContext->mScalingMode = *((uint32_t *)p->data + 1);
ALOGV("set mScalingMode = %d", pContext->mScalingMode);
break;
+ case VISUALIZER_PARAM_LATENCY:
+ pContext->mLatency = *((uint32_t *)p->data + 1);
+ ALOGV("set mLatency = %d", pContext->mLatency);
+ break;
default:
*(int32_t *)pReplyData = -EINVAL;
}
@@ -481,13 +486,9 @@
return -EINVAL;
}
if (pContext->mState == VISUALIZER_STATE_ACTIVE) {
- memcpy(pReplyData,
- pContext->mCaptureBuf[pContext->mCurrentBuf ^ 1],
- pContext->mCaptureSize);
- // if audio framework has stopped playing audio although the effect is still
- // active we must clear the capture buffer to return silence
- if ((pContext->mLastBuf == pContext->mCurrentBuf) &&
- (pContext->mBufferUpdateTime.tv_sec != 0)) {
+ int32_t latencyMs = pContext->mLatency;
+ uint32_t deltaMs = 0;
+ if (pContext->mBufferUpdateTime.tv_sec != 0) {
struct timespec ts;
if (clock_gettime(CLOCK_MONOTONIC, &ts) == 0) {
time_t secs = ts.tv_sec - pContext->mBufferUpdateTime.tv_sec;
@@ -496,17 +497,45 @@
--secs;
nsec += 1000000000;
}
- uint32_t deltaMs = secs * 1000 + nsec / 1000000;
- if (deltaMs > MAX_STALL_TIME_MS) {
- ALOGV("capture going to idle");
- pContext->mBufferUpdateTime.tv_sec = 0;
- memset(pContext->mCaptureBuf[pContext->mCurrentBuf ^ 1],
- 0x80,
- pContext->mCaptureSize);
+ deltaMs = secs * 1000 + nsec / 1000000;
+ latencyMs -= deltaMs;
+ if (latencyMs < 0) {
+ latencyMs = 0;
}
}
}
- pContext->mLastBuf = pContext->mCurrentBuf;
+ uint32_t deltaSmpl = pContext->mConfig.inputCfg.samplingRate * latencyMs / 1000;
+
+ int32_t capturePoint = pContext->mCaptureIdx - pContext->mCaptureSize - deltaSmpl;
+ int32_t captureSize = pContext->mCaptureSize;
+ if (capturePoint < 0) {
+ int32_t size = -capturePoint;
+ if (size > captureSize) {
+ size = captureSize;
+ }
+ memcpy(pReplyData,
+ pContext->mCaptureBuf + CAPTURE_BUF_SIZE + capturePoint,
+ size);
+ pReplyData += size;
+ captureSize -= size;
+ capturePoint = 0;
+ }
+ memcpy(pReplyData,
+ pContext->mCaptureBuf + capturePoint,
+ captureSize);
+
+
+ // if audio framework has stopped playing audio although the effect is still
+ // active we must clear the capture buffer to return silence
+ if ((pContext->mLastCaptureIdx == pContext->mCaptureIdx) &&
+ (pContext->mBufferUpdateTime.tv_sec != 0)) {
+ if (deltaMs > MAX_STALL_TIME_MS) {
+ ALOGV("capture going to idle");
+ pContext->mBufferUpdateTime.tv_sec = 0;
+ memset(pReplyData, 0x80, pContext->mCaptureSize);
+ }
+ }
+ pContext->mLastCaptureIdx = pContext->mCaptureIdx;
} else {
memset(pReplyData, 0x80, pContext->mCaptureSize);
}
diff --git a/media/libmedia/AudioParameter.cpp b/media/libmedia/AudioParameter.cpp
index 9766ee6..e3fea77 100644
--- a/media/libmedia/AudioParameter.cpp
+++ b/media/libmedia/AudioParameter.cpp
@@ -31,6 +31,7 @@
const char * const AudioParameter::keyChannels = AUDIO_PARAMETER_STREAM_CHANNELS;
const char * const AudioParameter::keyFrameCount = AUDIO_PARAMETER_STREAM_FRAME_COUNT;
const char * const AudioParameter::keyInputSource = AUDIO_PARAMETER_STREAM_INPUT_SOURCE;
+const char * const AudioParameter::keyScreenState = AUDIO_PARAMETER_KEY_SCREEN_STATE;
AudioParameter::AudioParameter(const String8& keyValuePairs)
{
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 14b7fc1..ee843aa 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -30,6 +30,9 @@
#LOCAL_C_INCLUDES += path/to/libsndfile/src
#LOCAL_STATIC_LIBRARIES += libsndfile
+# uncomment for systrace
+# LOCAL_CFLAGS += -DATRACE_TAG=ATRACE_TAG_AUDIO
+
LOCAL_MODULE := libnbaio
include $(BUILD_STATIC_LIBRARY)
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 2cfc3e8..be59ca0 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -164,6 +164,9 @@
// up large writes into smaller ones, and the wrapper would need to deal with scheduler.
} kUseFastMixer = FastMixer_Static;
+static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off"
+ // AudioFlinger::setParameters() updates, other threads read w/o lock
+
// ----------------------------------------------------------------------------
#ifdef ADD_BATTERY_DATA
@@ -889,6 +892,13 @@
mBtNrecIsOff = btNrecIsOff;
}
}
+ String8 screenState;
+ if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
+ bool isOff = screenState == "off";
+ if (isOff != (gScreenState & 1)) {
+ gScreenState = ((gScreenState & ~1) + 2) | isOff;
+ }
+ }
return final_result;
}
@@ -1501,6 +1511,7 @@
mMixerStatus(MIXER_IDLE),
mMixerStatusIgnoringFastTracks(MIXER_IDLE),
standbyDelay(AudioFlinger::mStandbyTimeInNsecs),
+ mScreenState(gScreenState),
// index 0 is reserved for normal mixer's submix
mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1)
{
@@ -1818,6 +1829,10 @@
uint32_t AudioFlinger::PlaybackThread::latency() const
{
Mutex::Autolock _l(mLock);
+ return latency_l();
+}
+uint32_t AudioFlinger::PlaybackThread::latency_l() const
+{
if (initCheck() == NO_ERROR) {
return correctLatency(mOutput->stream->get_latency(mOutput->stream));
} else {
@@ -2220,6 +2235,8 @@
size_t numCounterOffers = 0;
ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
+ monoPipe->setAvgFrames((mScreenState & 1) ?
+ (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
mPipeSink = monoPipe;
#ifdef TEE_SINK_FRAMES
@@ -2682,6 +2699,16 @@
#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
Tracer::traceBegin(ATRACE_TAG, "write");
#endif
+ // update the setpoint when gScreenState changes
+ uint32_t screenState = gScreenState;
+ if (screenState != mScreenState) {
+ mScreenState = screenState;
+ MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
+ if (pipe != NULL) {
+ pipe->setAvgFrames((mScreenState & 1) ?
+ (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
+ }
+ }
ssize_t framesWritten = mNormalSink->write(mMixBuffer, count);
#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER)
Tracer::traceEnd(ATRACE_TAG);
@@ -3887,6 +3914,7 @@
}
sleepTime = 0;
writeFrames = mNormalFrameCount;
+ standbyTime = systemTime() + standbyDelay;
}
void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
@@ -3898,21 +3926,19 @@
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0) {
- // flush remaining overflow buffers in output tracks
- for (size_t i = 0; i < outputTracks.size(); i++) {
- if (outputTracks[i]->isActive()) {
- sleepTime = 0;
- writeFrames = 0;
- memset(mMixBuffer, 0, mixBufferSize);
- break;
- }
+ if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+ writeFrames = mNormalFrameCount;
+ memset(mMixBuffer, 0, mixBufferSize);
+ } else {
+ // flush remaining overflow buffers in output tracks
+ writeFrames = 0;
}
+ sleepTime = 0;
}
}
void AudioFlinger::DuplicatingThread::threadLoop_write()
{
- standbyTime = systemTime() + standbyDelay;
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->write(mMixBuffer, writeFrames);
}
@@ -8187,6 +8213,31 @@
status = cmdStatus;
}
+ if (status == 0 &&
+ (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) {
+ uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
+ effect_param_t *p = (effect_param_t *)buf32;
+
+ p->psize = sizeof(uint32_t);
+ p->vsize = sizeof(uint32_t);
+ size = sizeof(int);
+ *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY;
+
+ uint32_t latency = 0;
+ PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId);
+ if (pbt != NULL) {
+ latency = pbt->latency_l();
+ }
+
+ *((int32_t *)p->data + 1)= latency;
+ (*mEffectInterface)->command(mEffectInterface,
+ EFFECT_CMD_SET_PARAM,
+ sizeof(effect_param_t) + 8,
+ &buf32,
+ &size,
+ &cmdStatus);
+ }
+
mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
(1000 * mConfig.outputCfg.buffer.frameCount);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index a0e0ea5..677d466 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -965,6 +965,8 @@
// return estimated latency in milliseconds, as reported by HAL
uint32_t latency() const;
+ // same, but lock must already be held
+ uint32_t latency_l() const;
void setMasterVolume(float value);
void setMasterMute(bool muted);
@@ -1117,6 +1119,7 @@
// For dumpsys
sp<NBAIO_Sink> mTeeSink;
sp<NBAIO_Source> mTeeSource;
+ uint32_t mScreenState; // cached copy of gScreenState
public:
virtual bool hasFastMixer() const = 0;
virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index d8bed40..3bb7b44 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -483,7 +483,10 @@
(int) sec, nsec / 1000000L);
dumpState->mOverruns++;
}
- sleepNs = periodNs - overrunNs;
+ // Code for non blocking audio HAL. Sleep time must be tuned to allow
+ // catching up after an underrun
+ // sleepNs = periodNs - overrunNs;
+ sleepNs = -1;
} else {
sleepNs = -1;
ignoreNextOverrun = false;
diff --git a/services/audioflinger/MonoPipe.cpp b/services/audioflinger/MonoPipe.cpp
index 6efb8b1..f3fc19a 100644
--- a/services/audioflinger/MonoPipe.cpp
+++ b/services/audioflinger/MonoPipe.cpp
@@ -20,6 +20,7 @@
#include <cutils/atomic.h>
#include <cutils/compiler.h>
#include <utils/Log.h>
+#include <utils/Trace.h>
#include "MonoPipe.h"
#include "roundup.h"
@@ -32,6 +33,9 @@
mBuffer(malloc(mMaxFrames * Format_frameSize(format))),
mFront(0),
mRear(0),
+ mWriteTsValid(false),
+ // mWriteTs
+ mSetpoint((reqFrames * 11) / 16),
mWriteCanBlock(writeCanBlock)
{
}
@@ -87,40 +91,75 @@
count -= written;
buffer = (char *) buffer + (written << mBitShift);
// Simulate blocking I/O by sleeping at different rates, depending on a throttle.
- // The throttle tries to keep the pipe about 11/16 full on average, with a slight jitter.
+ // The throttle tries to keep the mean pipe depth near the setpoint, with a slight jitter.
uint32_t ns;
if (written > 0) {
size_t filled = (mMaxFrames - avail) + written;
// FIXME cache these values to avoid re-computation
- if (filled <= mReqFrames / 4) {
+ if (filled <= mSetpoint / 2) {
// pipe is (nearly) empty, fill quickly
ns = written * ( 500000000 / Format_sampleRate(mFormat));
- } else if (filled <= mReqFrames / 2) {
- // pipe is normal, fill at slightly faster rate
+ } else if (filled <= (mSetpoint * 3) / 4) {
+ // pipe is below setpoint, fill at slightly faster rate
ns = written * ( 750000000 / Format_sampleRate(mFormat));
- } else if (filled <= (mReqFrames * 5) / 8) {
- // pipe is normal, fill at nominal rate
+ } else if (filled <= (mSetpoint * 5) / 4) {
+ // pipe is at setpoint, fill at nominal rate
ns = written * (1000000000 / Format_sampleRate(mFormat));
- } else if (filled <= (mReqFrames * 3) / 4) {
- // pipe is normal, fill at slightly slower rate
- ns = written * (1100000000 / Format_sampleRate(mFormat));
+ } else if (filled <= (mSetpoint * 3) / 2) {
+ // pipe is above setpoint, fill at slightly slower rate
+ ns = written * (1150000000 / Format_sampleRate(mFormat));
+ } else if (filled <= (mSetpoint * 7) / 4) {
+ // pipe is overflowing, fill slowly
+ ns = written * (1350000000 / Format_sampleRate(mFormat));
} else {
- // pipe is (nearly) full, fill slowly
- ns = written * (1250000000 / Format_sampleRate(mFormat));
+ // pipe is severely overflowing
+ ns = written * (1750000000 / Format_sampleRate(mFormat));
}
} else {
- ns = mReqFrames * (250000000 / Format_sampleRate(mFormat));
+ ns = count * (1350000000 / Format_sampleRate(mFormat));
}
if (ns > 999999999) {
ns = 999999999;
}
- struct timespec sleep;
- sleep.tv_sec = 0;
- sleep.tv_nsec = ns;
- nanosleep(&sleep, NULL);
+ struct timespec nowTs;
+ bool nowTsValid = !clock_gettime(CLOCK_MONOTONIC, &nowTs);
+ // deduct the elapsed time since previous write() completed
+ if (nowTsValid && mWriteTsValid) {
+ time_t sec = nowTs.tv_sec - mWriteTs.tv_sec;
+ long nsec = nowTs.tv_nsec - mWriteTs.tv_nsec;
+ if (nsec < 0) {
+ --sec;
+ nsec += 1000000000;
+ }
+ if (sec == 0) {
+ if ((long) ns > nsec) {
+ ns -= nsec;
+ } else {
+ ns = 0;
+ }
+ }
+ }
+ if (ns > 0) {
+ const struct timespec req = {0, ns};
+ nanosleep(&req, NULL);
+ }
+ // record the time that this write() completed
+ if (nowTsValid) {
+ mWriteTs = nowTs;
+ if ((mWriteTs.tv_nsec += ns) >= 1000000000) {
+ mWriteTs.tv_nsec -= 1000000000;
+ ++mWriteTs.tv_sec;
+ }
+ }
+ mWriteTsValid = nowTsValid;
}
mFramesWritten += totalFramesWritten;
return totalFramesWritten;
}
+void MonoPipe::setAvgFrames(size_t setpoint)
+{
+ mSetpoint = setpoint;
+}
+
} // namespace android
diff --git a/services/audioflinger/MonoPipe.h b/services/audioflinger/MonoPipe.h
index aaaa51f..f6e2cb3 100644
--- a/services/audioflinger/MonoPipe.h
+++ b/services/audioflinger/MonoPipe.h
@@ -58,7 +58,9 @@
// average number of frames present in the pipe under normal conditions.
// See throttling mechanism in MonoPipe::write()
- size_t getAvgFrames() const { return (mReqFrames * 11) / 16; }
+ size_t getAvgFrames() const { return mSetpoint; }
+ void setAvgFrames(size_t setpoint);
+ size_t maxFrames() const { return mMaxFrames; }
private:
const size_t mReqFrames; // as requested in constructor, unrounded
@@ -71,6 +73,9 @@
// read by writer with android_atomic_acquire_load
volatile int32_t mRear; // written by writer with android_atomic_release_store,
// read by reader with android_atomic_acquire_load
+ bool mWriteTsValid; // whether mWriteTs is valid
+ struct timespec mWriteTs; // time that the previous write() completed
+ size_t mSetpoint; // target value for pipe fill depth
const bool mWriteCanBlock; // whether write() should block if the pipe is full
};