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-rw-r--r--services/audioflinger/AudioFlinger.cpp22
-rw-r--r--services/audioflinger/AudioMixer.cpp3
-rw-r--r--services/audioflinger/AudioMixer.h28
3 files changed, 32 insertions, 21 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 0292bc3..2e2834c 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -2295,7 +2295,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
uint16_t sendLevel = cblk->getSendLevel_U4_12();
// send level comes from shared memory and so may be corrupt
- if (sendLevel >= MAX_GAIN_INT) {
+ if (sendLevel > MAX_GAIN_INT) {
ALOGV("Track send level out of range: %04X", sendLevel);
sendLevel = MAX_GAIN_INT;
}
@@ -2316,25 +2316,21 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
}
// Convert volumes from 8.24 to 4.12 format
- int16_t left, right, aux;
// This additional clamping is needed in case chain->setVolume_l() overshot
- uint32_t v_clamped = (vl + (1 << 11)) >> 12;
- if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
- left = int16_t(v_clamped);
- v_clamped = (vr + (1 << 11)) >> 12;
- if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
- right = int16_t(v_clamped);
+ vl = (vl + (1 << 11)) >> 12;
+ if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT;
+ vr = (vr + (1 << 11)) >> 12;
+ if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT;
- if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
- aux = int16_t(va);
+ if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for -
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(name, track);
mAudioMixer->enable(name);
- 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, param, AudioMixer::VOLUME0, (void *)vl);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
+ mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index f3bf953..020d62a 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -257,6 +257,7 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
}
break;
case AUXLEVEL:
+ //assert(0 <= valueInt && valueInt <= MAX_GAIN_INT);
if (track.auxLevel != valueInt) {
ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
track.prevAuxLevel = track.auxLevel << 16;
@@ -565,7 +566,7 @@ void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32
const int16_t vr = t->volume[1];
if (CC_UNLIKELY(aux != NULL)) {
- const int16_t va = (int16_t)t->auxLevel;
+ const int16_t va = t->auxLevel;
do {
int16_t l = (int16_t)(*temp++ >> 12);
int16_t r = (int16_t)(*temp++ >> 12);
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index c3174ae..b210212 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -127,32 +127,46 @@ private:
int32_t prevVolume[MAX_NUM_CHANNELS];
+ // 16-byte boundary
+
int32_t volumeInc[MAX_NUM_CHANNELS];
- int32_t auxLevel;
int32_t auxInc;
int32_t prevAuxLevel;
+ // 16-byte boundary
+
+ int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
uint16_t frameCount;
- uint8_t channelCount : 4;
- uint8_t enabled : 1;
- uint8_t reserved0 : 3;
- uint8_t format;
- uint32_t channelMask;
+ uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
+ uint8_t format; // always 16
+ uint16_t enabled; // actually bool
+ uint32_t channelMask; // currently under-used
AudioBufferProvider* bufferProvider;
- mutable AudioBufferProvider::Buffer buffer;
+
+ // 16-byte boundary
+
+ mutable AudioBufferProvider::Buffer buffer; // 8 bytes
hook_t hook;
const void* in; // current location in buffer
+ // 16-byte boundary
+
AudioResampler* resampler;
uint32_t sampleRate;
int32_t* mainBuffer;
int32_t* auxBuffer;
+ // 16-byte boundary
+
uint64_t localTimeFreq;
+ int64_t padding;
+
+ // 16-byte boundary
+
bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
bool doesResample() const { return resampler != NULL; }
void resetResampler() { if (resampler != NULL) resampler->reset(); }