Merge "Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE"
diff --git a/media/libeffects/testlibs/AudioBiquadFilter.cpp b/media/libeffects/testlibs/AudioBiquadFilter.cpp
index 72917a3..16dd1c5 100644
--- a/media/libeffects/testlibs/AudioBiquadFilter.cpp
+++ b/media/libeffects/testlibs/AudioBiquadFilter.cpp
@@ -17,12 +17,10 @@
#include <string.h>
#include <assert.h>
+#include <cutils/compiler.h>
#include "AudioBiquadFilter.h"
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
-
namespace android {
const audio_coef_t AudioBiquadFilter::IDENTITY_COEFS[AudioBiquadFilter::NUM_COEFS] = { AUDIO_COEF_ONE, 0, 0, 0, 0 };
@@ -55,7 +53,7 @@
void AudioBiquadFilter::setCoefs(const audio_coef_t coefs[NUM_COEFS], bool immediate) {
memcpy(mTargetCoefs, coefs, sizeof(mTargetCoefs));
if (mState & STATE_ENABLED_MASK) {
- if (UNLIKELY(immediate)) {
+ if (CC_UNLIKELY(immediate)) {
memcpy(mCoefs, coefs, sizeof(mCoefs));
setState(STATE_NORMAL);
} else {
@@ -70,7 +68,7 @@
}
void AudioBiquadFilter::enable(bool immediate) {
- if (UNLIKELY(immediate)) {
+ if (CC_UNLIKELY(immediate)) {
memcpy(mCoefs, mTargetCoefs, sizeof(mCoefs));
setState(STATE_NORMAL);
} else {
@@ -79,7 +77,7 @@
}
void AudioBiquadFilter::disable(bool immediate) {
- if (UNLIKELY(immediate)) {
+ if (CC_UNLIKELY(immediate)) {
memcpy(mCoefs, IDENTITY_COEFS, sizeof(mCoefs));
setState(STATE_BYPASS);
} else {
@@ -142,7 +140,7 @@
audio_sample_t * out,
int frameCount) {
// The common case is in-place processing, because this is what the EQ does.
- if (UNLIKELY(in != out)) {
+ if (CC_UNLIKELY(in != out)) {
memcpy(out, in, frameCount * mNumChannels * sizeof(audio_sample_t));
}
}
diff --git a/media/libeffects/testlibs/AudioCoefInterpolator.cpp b/media/libeffects/testlibs/AudioCoefInterpolator.cpp
index 039ab9f..6b56922 100644
--- a/media/libeffects/testlibs/AudioCoefInterpolator.cpp
+++ b/media/libeffects/testlibs/AudioCoefInterpolator.cpp
@@ -16,10 +16,10 @@
*/
#include <string.h>
-#include "AudioCoefInterpolator.h"
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
+
+#include "AudioCoefInterpolator.h"
namespace android {
@@ -44,9 +44,9 @@
size_t index = 0;
size_t dim = mNumInDims;
while (dim-- > 0) {
- if (UNLIKELY(intCoord[dim] < 0)) {
+ if (CC_UNLIKELY(intCoord[dim] < 0)) {
fracCoord[dim] = 0;
- } else if (UNLIKELY(intCoord[dim] >= (int)mInDims[dim] - 1)) {
+ } else if (CC_UNLIKELY(intCoord[dim] >= (int)mInDims[dim] - 1)) {
fracCoord[dim] = 0;
index += mInDimOffsets[dim] * (mInDims[dim] - 1);
} else {
@@ -63,7 +63,7 @@
memcpy(out, mTable + index, mNumOutDims * sizeof(audio_coef_t));
} else {
getCoefRecurse(index, fracCoord, out, dim + 1);
- if (LIKELY(fracCoord != 0)) {
+ if (CC_LIKELY(fracCoord != 0)) {
audio_coef_t tempCoef[MAX_OUT_DIMS];
getCoefRecurse(index + mInDimOffsets[dim], fracCoord, tempCoef,
dim + 1);
diff --git a/media/libeffects/testlibs/AudioCommon.h b/media/libeffects/testlibs/AudioCommon.h
index 444f93a..e8080dc 100644
--- a/media/libeffects/testlibs/AudioCommon.h
+++ b/media/libeffects/testlibs/AudioCommon.h
@@ -20,6 +20,7 @@
#include <stdint.h>
#include <stddef.h>
+#include <cutils/compiler.h>
namespace android {
@@ -76,9 +77,9 @@
// Convert a audio_sample_t sample to S15 (with clipping)
inline int16_t audio_sample_t_to_s15_clip(audio_sample_t sample) {
// TODO: optimize for targets supporting this as an atomic operation.
- if (__builtin_expect(sample >= (0x7FFF << 9), 0)) {
+ if (CC_UNLIKELY(sample >= (0x7FFF << 9))) {
return 0x7FFF;
- } else if (__builtin_expect(sample <= -(0x8000 << 9), 0)) {
+ } else if (CC_UNLIKELY(sample <= -(0x8000 << 9))) {
return 0x8000;
} else {
return audio_sample_t_to_s15(sample);
diff --git a/media/libeffects/testlibs/AudioPeakingFilter.cpp b/media/libeffects/testlibs/AudioPeakingFilter.cpp
index 60fefe6..99323ac 100644
--- a/media/libeffects/testlibs/AudioPeakingFilter.cpp
+++ b/media/libeffects/testlibs/AudioPeakingFilter.cpp
@@ -21,9 +21,7 @@
#include <new>
#include <assert.h>
-
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
namespace android {
// Format of the coefficient table:
@@ -66,12 +64,12 @@
void AudioPeakingFilter::setFrequency(uint32_t millihertz) {
mNominalFrequency = millihertz;
- if (UNLIKELY(millihertz > mNiquistFreq / 2)) {
+ if (CC_UNLIKELY(millihertz > mNiquistFreq / 2)) {
millihertz = mNiquistFreq / 2;
}
uint32_t normFreq = static_cast<uint32_t>(
(static_cast<uint64_t>(millihertz) * mFrequencyFactor) >> 10);
- if (LIKELY(normFreq > (1 << 23))) {
+ if (CC_LIKELY(normFreq > (1 << 23))) {
mFrequency = (Effects_log2(normFreq) - ((32-9) << 15)) << (FREQ_PRECISION_BITS - 15);
} else {
mFrequency = 0;
@@ -107,11 +105,11 @@
int32_t halfBW = (((mBandwidth + 1) / 2) << 15) / 1200;
low = static_cast<uint32_t>((static_cast<uint64_t>(mNominalFrequency) * Effects_exp2(-halfBW + (16 << 15))) >> 16);
- if (UNLIKELY(halfBW >= (16 << 15))) {
+ if (CC_UNLIKELY(halfBW >= (16 << 15))) {
high = mNiquistFreq;
} else {
high = static_cast<uint32_t>((static_cast<uint64_t>(mNominalFrequency) * Effects_exp2(halfBW + (16 << 15))) >> 16);
- if (UNLIKELY(high > mNiquistFreq)) {
+ if (CC_UNLIKELY(high > mNiquistFreq)) {
high = mNiquistFreq;
}
}
diff --git a/media/libeffects/testlibs/AudioShelvingFilter.cpp b/media/libeffects/testlibs/AudioShelvingFilter.cpp
index b8650ba..e031287 100644
--- a/media/libeffects/testlibs/AudioShelvingFilter.cpp
+++ b/media/libeffects/testlibs/AudioShelvingFilter.cpp
@@ -21,9 +21,7 @@
#include <new>
#include <assert.h>
-
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
namespace android {
// Format of the coefficient tables:
@@ -71,13 +69,13 @@
void AudioShelvingFilter::setFrequency(uint32_t millihertz) {
mNominalFrequency = millihertz;
- if (UNLIKELY(millihertz > mNiquistFreq / 2)) {
+ if (CC_UNLIKELY(millihertz > mNiquistFreq / 2)) {
millihertz = mNiquistFreq / 2;
}
uint32_t normFreq = static_cast<uint32_t>(
(static_cast<uint64_t>(millihertz) * mFrequencyFactor) >> 10);
uint32_t log2minFreq = (mType == kLowShelf ? (32-10) : (32-2));
- if (LIKELY(normFreq > (1U << log2minFreq))) {
+ if (CC_LIKELY(normFreq > (1U << log2minFreq))) {
mFrequency = (Effects_log2(normFreq) - (log2minFreq << 15)) << (FREQ_PRECISION_BITS - 15);
} else {
mFrequency = 0;
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 1d8e15b..dd1c6a5 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -39,9 +39,7 @@
#include <system/audio.h>
#include <cutils/bitops.h>
-
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
namespace android {
// ---------------------------------------------------------------------------
@@ -513,11 +511,11 @@
goto start_loop_here;
while (framesReady == 0) {
active = mActive;
- if (UNLIKELY(!active)) {
+ if (CC_UNLIKELY(!active)) {
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
- if (UNLIKELY(!waitCount)) {
+ if (CC_UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
@@ -534,7 +532,7 @@
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_record;
}
- if (__builtin_expect(result!=NO_ERROR, false)) {
+ if (CC_UNLIKELY(result != NO_ERROR)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
LOGW( "obtainBuffer timed out (is the CPU pegged?) "
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 6868d99..e0c6ca5 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -38,13 +38,11 @@
#include <utils/Atomic.h>
#include <cutils/bitops.h>
+#include <cutils/compiler.h>
#include <system/audio.h>
#include <system/audio_policy.h>
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
-
namespace android {
// ---------------------------------------------------------------------------
@@ -856,12 +854,12 @@
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
- if (UNLIKELY(!active)) {
+ if (CC_UNLIKELY(!active)) {
ALOGV("Not active and NO_MORE_BUFFERS");
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
- if (UNLIKELY(!waitCount)) {
+ if (CC_UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
@@ -879,7 +877,7 @@
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_track;
}
- if (__builtin_expect(result!=NO_ERROR, false)) {
+ if (CC_UNLIKELY(result != NO_ERROR)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
// timing out when a loop has been set and we have already written upto loop end
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index a893cf2..ab94bb8 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -35,6 +35,7 @@
#include <cutils/bitops.h>
#include <cutils/properties.h>
+#include <cutils/compiler.h>
#include <media/AudioTrack.h>
#include <media/AudioRecord.h>
@@ -1930,8 +1931,8 @@
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
// put audio hardware into standby after short delay
- if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
- mSuspended) {
+ if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+ mSuspended)) {
if (!mStandby) {
ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
mOutput->stream->common.standby(&mOutput->stream->common);
@@ -1976,7 +1977,7 @@
lockEffectChains_l(effectChains);
}
- if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+ if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
mAudioMixer->process();
sleepTime = 0;
@@ -2101,12 +2102,13 @@
sp<Track> t = activeTracks[i].promote();
if (t == 0) continue;
+ // this const just means the local variable doesn't change
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// The first time a track is added we wait
// for all its buffers to be filled before processing it
- mAudioMixer->setActiveTrack(track->name());
+ int name = track->name();
// make sure that we have enough frames to mix one full buffer.
// enforce this condition only once to enable draining the buffer in case the client
// app does not call stop() and relies on underrun to stop:
@@ -2118,13 +2120,21 @@
if (t->sampleRate() == (int)mSampleRate) {
minFrames = mFrameCount;
} else {
- minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1;
+ // +1 for rounding and +1 for additional sample needed for interpolation
+ minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
+ // add frames already consumed but not yet released by the resampler
+ // because cblk->framesReady() will include these frames
+ minFrames += mAudioMixer->getUnreleasedFrames(track->name());
+ // the minimum track buffer size is normally twice the number of frames necessary
+ // to fill one buffer and the resampler should not leave more than one buffer worth
+ // of unreleased frames after each pass, but just in case...
+ LOG_ASSERT(minFrames <= cblk->frameCount);
}
}
if ((cblk->framesReady() >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
- //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
+ //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this);
mixedTracks++;
@@ -2137,8 +2147,8 @@
if (chain != 0) {
tracksWithEffect++;
} else {
- LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
- track->name(), track->sessionId());
+ LOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
+ name, track->sessionId());
}
}
@@ -2151,7 +2161,7 @@
track->mState = TrackBase::ACTIVE;
param = AudioMixer::RAMP_VOLUME;
}
- mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
+ mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
} else if (cblk->server != 0) {
// If the track is stopped before the first frame was mixed,
// do not apply ramp
@@ -2203,26 +2213,31 @@
aux = int16_t(va);
// XXX: these things DON'T need to be done each time
- mAudioMixer->setBufferProvider(track);
- mAudioMixer->enable();
+ mAudioMixer->setBufferProvider(name, track);
+ mAudioMixer->enable(name);
- mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
- mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
- mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right);
+ mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux);
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::FORMAT, (void *)track->format());
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
mAudioMixer->setParameter(
+ name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
(void *)(cblk->sampleRate));
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
@@ -2230,7 +2245,7 @@
track->mRetryCount = kMaxTrackRetries;
mixerStatus = MIXER_TRACKS_READY;
} else {
- //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
+ //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
if (track->isStopped()) {
track->reset();
}
@@ -2242,7 +2257,7 @@
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->mRetryCount) <= 0) {
- ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
+ ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
tracksToRemove->add(track);
// indicate to client process that the track was disabled because of underrun
android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
@@ -2250,13 +2265,13 @@
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
- mAudioMixer->disable();
+ mAudioMixer->disable(name);
}
}
// remove all the tracks that need to be...
count = tracksToRemove->size();
- if (UNLIKELY(count)) {
+ if (CC_UNLIKELY(count)) {
for (size_t i=0 ; i<count ; i++) {
const sp<Track>& track = tracksToRemove->itemAt(i);
mActiveTracks.remove(track);
@@ -2591,8 +2606,8 @@
}
// put audio hardware into standby after short delay
- if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
- mSuspended) {
+ if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
+ mSuspended)) {
// wait until we have something to do...
if (!mStandby) {
ALOGV("Audio hardware entering standby, mixer %p\n", this);
@@ -2742,7 +2757,7 @@
}
// remove all the tracks that need to be...
- if (UNLIKELY(trackToRemove != 0)) {
+ if (CC_UNLIKELY(trackToRemove != 0)) {
mActiveTracks.remove(trackToRemove);
if (!effectChains.isEmpty()) {
ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
@@ -2757,7 +2772,7 @@
lockEffectChains_l(effectChains);
}
- if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+ if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
AudioBufferProvider::Buffer buffer;
size_t frameCount = mFrameCount;
curBuf = (int8_t *)mMixBuffer;
@@ -2765,7 +2780,7 @@
while (frameCount) {
buffer.frameCount = frameCount;
activeTrack->getNextBuffer(&buffer);
- if (UNLIKELY(buffer.raw == NULL)) {
+ if (CC_UNLIKELY(buffer.raw == NULL)) {
memset(curBuf, 0, frameCount * mFrameSize);
break;
}
@@ -2984,8 +2999,8 @@
}
// put audio hardware into standby after short delay
- if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
- mSuspended) {
+ if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+ mSuspended)) {
if (!mStandby) {
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->stop();
@@ -3030,7 +3045,7 @@
lockEffectChains_l(effectChains);
}
- if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+ if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
if (outputsReady(outputTracks)) {
mAudioMixer->process();
@@ -3443,7 +3458,7 @@
framesReady = cblk->framesReady();
- if (LIKELY(framesReady)) {
+ if (CC_LIKELY(framesReady)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
@@ -3692,7 +3707,7 @@
framesAvail = cblk->framesAvailable_l();
- if (LIKELY(framesAvail)) {
+ if (CC_LIKELY(framesAvail)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
@@ -3945,7 +3960,7 @@
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
- if (UNLIKELY(!active)) {
+ if (CC_UNLIKELY(!active)) {
ALOGV("Not active and NO_MORE_BUFFERS");
return AudioTrack::NO_MORE_BUFFERS;
}
@@ -4323,7 +4338,7 @@
}
buffer.frameCount = mFrameCount;
- if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+ if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
size_t framesOut = buffer.frameCount;
if (mResampler == NULL) {
// no resampling
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 9707cf4..7baa8fc 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -55,12 +55,6 @@
// ----------------------------------------------------------------------------
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
-
-
-// ----------------------------------------------------------------------------
-
static const nsecs_t kStandbyTimeInNsecs = seconds(3);
class AudioFlinger :
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index dc1d1a7..c9c61a5 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -28,6 +28,7 @@
#include <utils/Log.h>
#include <cutils/bitops.h>
+#include <cutils/compiler.h>
#include <system/audio.h>
@@ -40,7 +41,7 @@
// ----------------------------------------------------------------------------
AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
- : mActiveTrack(0), mTrackNames(0), mSampleRate(sampleRate)
+ : mTrackNames(0), mSampleRate(sampleRate)
{
// AudioMixer is not yet capable of multi-channel beyond stereo
assert(2 == MAX_NUM_CHANNELS);
@@ -139,120 +140,120 @@
mTrackNames &= ~(1<<name);
}
-void AudioMixer::enable()
+void AudioMixer::enable(int name)
{
- if (mState.tracks[ mActiveTrack ].enabled != 1) {
- mState.tracks[ mActiveTrack ].enabled = 1;
- ALOGV("enable(%d)", mActiveTrack);
- invalidateState(1<<mActiveTrack);
+ name -= TRACK0;
+ assert(uint32_t(name) < MAX_NUM_TRACKS);
+ track_t& track = mState.tracks[name];
+
+ if (track.enabled != 1) {
+ track.enabled = 1;
+ ALOGV("enable(%d)", name);
+ invalidateState(1 << name);
}
}
-void AudioMixer::disable()
+void AudioMixer::disable(int name)
{
- if (mState.tracks[ mActiveTrack ].enabled != 0) {
- mState.tracks[ mActiveTrack ].enabled = 0;
- ALOGV("disable(%d)", mActiveTrack);
- invalidateState(1<<mActiveTrack);
+ name -= TRACK0;
+ assert(uint32_t(name) < MAX_NUM_TRACKS);
+ track_t& track = mState.tracks[name];
+
+ if (track.enabled != 0) {
+ track.enabled = 0;
+ ALOGV("disable(%d)", name);
+ invalidateState(1 << name);
}
}
-void AudioMixer::setActiveTrack(int track)
+void AudioMixer::setParameter(int name, int target, int param, void *value)
{
- // this also catches track < TRACK0
- track -= TRACK0;
- assert(uint32_t(track) < MAX_NUM_TRACKS);
- mActiveTrack = track;
-}
+ name -= TRACK0;
+ assert(uint32_t(name) < MAX_NUM_TRACKS);
+ track_t& track = mState.tracks[name];
-void AudioMixer::setParameter(int target, int name, void *value)
-{
int valueInt = (int)value;
int32_t *valueBuf = (int32_t *)value;
switch (target) {
case TRACK:
- switch (name) {
+ switch (param) {
case CHANNEL_MASK: {
uint32_t mask = (uint32_t)value;
- if (mState.tracks[ mActiveTrack ].channelMask != mask) {
+ if (track.channelMask != mask) {
uint8_t channelCount = popcount(mask);
assert((channelCount <= MAX_NUM_CHANNELS) && (channelCount));
- mState.tracks[ mActiveTrack ].channelMask = mask;
- mState.tracks[ mActiveTrack ].channelCount = channelCount;
+ track.channelMask = mask;
+ track.channelCount = channelCount;
ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
- invalidateState(1<<mActiveTrack);
+ invalidateState(1 << name);
}
} break;
case MAIN_BUFFER:
- if (mState.tracks[ mActiveTrack ].mainBuffer != valueBuf) {
- mState.tracks[ mActiveTrack ].mainBuffer = valueBuf;
+ if (track.mainBuffer != valueBuf) {
+ track.mainBuffer = valueBuf;
ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
- invalidateState(1<<mActiveTrack);
+ invalidateState(1 << name);
}
break;
case AUX_BUFFER:
- if (mState.tracks[ mActiveTrack ].auxBuffer != valueBuf) {
- mState.tracks[ mActiveTrack ].auxBuffer = valueBuf;
+ if (track.auxBuffer != valueBuf) {
+ track.auxBuffer = valueBuf;
ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
- invalidateState(1<<mActiveTrack);
+ invalidateState(1 << name);
}
break;
default:
- // bad name
+ // bad param
assert(false);
}
break;
case RESAMPLE:
- switch (name) {
- case SAMPLE_RATE: {
+ switch (param) {
+ case SAMPLE_RATE:
assert(valueInt > 0);
- track_t& track = mState.tracks[ mActiveTrack ];
if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
uint32_t(valueInt));
- invalidateState(1<<mActiveTrack);
+ invalidateState(1 << name);
}
- } break;
- case RESET: {
- track_t& track = mState.tracks[ mActiveTrack ];
+ break;
+ case RESET:
track.resetResampler();
- invalidateState(1<<mActiveTrack);
- } break;
+ invalidateState(1 << name);
+ break;
default:
- // bad name
+ // bad param
assert(false);
}
break;
case RAMP_VOLUME:
case VOLUME:
- switch (name) {
+ switch (param) {
case VOLUME0:
- case VOLUME1: {
- track_t& track = mState.tracks[ mActiveTrack ];
- if (track.volume[name-VOLUME0] != valueInt) {
+ case VOLUME1:
+ if (track.volume[param-VOLUME0] != valueInt) {
ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
- track.prevVolume[name-VOLUME0] = track.volume[name-VOLUME0] << 16;
- track.volume[name-VOLUME0] = valueInt;
+ track.prevVolume[param-VOLUME0] = track.volume[param-VOLUME0] << 16;
+ track.volume[param-VOLUME0] = valueInt;
if (target == VOLUME) {
- track.prevVolume[name-VOLUME0] = valueInt << 16;
- track.volumeInc[name-VOLUME0] = 0;
+ track.prevVolume[param-VOLUME0] = valueInt << 16;
+ track.volumeInc[param-VOLUME0] = 0;
} else {
- int32_t d = (valueInt<<16) - track.prevVolume[name-VOLUME0];
+ int32_t d = (valueInt<<16) - track.prevVolume[param-VOLUME0];
int32_t volInc = d / int32_t(mState.frameCount);
- track.volumeInc[name-VOLUME0] = volInc;
+ track.volumeInc[param-VOLUME0] = volInc;
if (volInc == 0) {
- track.prevVolume[name-VOLUME0] = valueInt << 16;
+ track.prevVolume[param-VOLUME0] = valueInt << 16;
}
}
- invalidateState(1<<mActiveTrack);
+ invalidateState(1 << name);
}
- } break;
- case AUXLEVEL: {
- track_t& track = mState.tracks[ mActiveTrack ];
+ break;
+ case AUXLEVEL:
if (track.auxLevel != valueInt) {
ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
track.prevAuxLevel = track.auxLevel << 16;
@@ -268,11 +269,11 @@
track.prevAuxLevel = valueInt << 16;
}
}
- invalidateState(1<<mActiveTrack);
+ invalidateState(1 << name);
}
- } break;
+ break;
default:
- // bad name
+ // bad param
assert(false);
}
break;
@@ -329,10 +330,29 @@
}
}
-
-void AudioMixer::setBufferProvider(AudioBufferProvider* buffer)
+size_t AudioMixer::track_t::getUnreleasedFrames()
{
- mState.tracks[ mActiveTrack ].bufferProvider = buffer;
+ if (resampler != NULL) {
+ return resampler->getUnreleasedFrames();
+ }
+ return 0;
+}
+
+size_t AudioMixer::getUnreleasedFrames(int name)
+{
+ name -= TRACK0;
+ if (uint32_t(name) < MAX_NUM_TRACKS) {
+ track_t& track(mState.tracks[name]);
+ return track.getUnreleasedFrames();
+ }
+ return 0;
+}
+
+void AudioMixer::setBufferProvider(int name, AudioBufferProvider* buffer)
+{
+ name -= TRACK0;
+ assert(uint32_t(name) < MAX_NUM_TRACKS);
+ mState.tracks[name].bufferProvider = buffer;
}
@@ -489,13 +509,13 @@
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
volumeRampStereo(t, out, outFrameCount, temp, aux);
} else {
volumeStereo(t, out, outFrameCount, temp, aux);
}
} else {
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
@@ -526,7 +546,7 @@
// (vl + vlInc*frameCount)/65536.0f, frameCount);
// ramp volume
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
int32_t va = t->prevAuxLevel;
const int32_t vaInc = t->auxInc;
int32_t l;
@@ -561,7 +581,7 @@
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
const int16_t va = (int16_t)t->auxLevel;
do {
int16_t l = (int16_t)(*temp++ >> 12);
@@ -588,11 +608,11 @@
{
int16_t const *in = static_cast<int16_t const *>(t->in);
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
int32_t l;
int32_t r;
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
@@ -637,7 +657,7 @@
}
} else {
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
@@ -678,9 +698,9 @@
{
int16_t const *in = static_cast<int16_t const *>(t->in);
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
@@ -723,7 +743,7 @@
}
} else {
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
@@ -776,7 +796,7 @@
i = 31 - __builtin_clz(e2);
e2 &= ~(1<<i);
track_t& t2 = state->tracks[i];
- if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<i);
}
}
@@ -834,7 +854,7 @@
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
- if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
@@ -851,7 +871,7 @@
track_t& t = state->tracks[i];
size_t outFrames = BLOCKSIZE;
int32_t *aux = NULL;
- if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+ if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
aux = t.auxBuffer + numFrames;
}
while (outFrames) {
@@ -860,7 +880,7 @@
(t.hook)(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux);
t.frameCount -= inFrames;
outFrames -= inFrames;
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
aux += inFrames;
}
}
@@ -915,7 +935,7 @@
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
- if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
@@ -927,7 +947,7 @@
e1 &= ~(1<<i);
track_t& t = state->tracks[i];
int32_t *aux = NULL;
- if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+ if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
aux = t.auxBuffer;
}
@@ -948,7 +968,7 @@
// been enabled for mixing.
if (t.in == NULL) break;
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
aux += outFrames;
}
(t.hook)(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux);
@@ -990,7 +1010,7 @@
}
size_t outFrames = b.frameCount;
- if (UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
+ if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
// volume is boosted, so we might need to clamp even though
// we process only one track.
do {
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 70464fc..4ba6845 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -28,11 +28,6 @@
// ----------------------------------------------------------------------------
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
-
-// ----------------------------------------------------------------------------
-
class AudioMixer
{
public:
@@ -47,7 +42,7 @@
enum { // names
- // track units (MAX_NUM_TRACKS units)
+ // track names (MAX_NUM_TRACKS units)
TRACK0 = 0x1000,
// 0x2000 is unused
@@ -74,20 +69,22 @@
};
+ // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
int getTrackName();
void deleteTrackName(int name);
- void enable();
- void disable();
+ void enable(int name);
+ void disable(int name);
- void setActiveTrack(int track);
- void setParameter(int target, int name, void *value);
+ void setParameter(int name, int target, int param, void *value);
- void setBufferProvider(AudioBufferProvider* bufferProvider);
+ void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
void process();
uint32_t trackNames() const { return mTrackNames; }
+ size_t getUnreleasedFrames(int name);
+
private:
enum {
@@ -159,6 +156,7 @@
bool doesResample() const;
void resetResampler();
void adjustVolumeRamp(bool aux);
+ size_t getUnreleasedFrames();
};
// pad to 32-bytes to fill cache line
@@ -173,7 +171,7 @@
track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
};
- int mActiveTrack;
+ // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
uint32_t mTrackNames;
const uint32_t mSampleRate;
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 9f06c1c..ffa690a 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -54,6 +54,7 @@
AudioBufferProvider* provider) = 0;
virtual void reset();
+ virtual size_t getUnreleasedFrames() { return mInputIndex; }
protected:
// number of bits for phase fraction - 30 bits allows nearly 2x downsampling